I used two sine waves on oscillators 1 & 2 and I used the orgn 3 wave on oscillator 3. I then transposed the first oscillator an octave higher. I used these oscillators because sine waves is just a pure signal and sine waves are really useful for providing depth and warmth to any sound. I did transpose the first oscillator an octave higher because I didn't want the first oscillator to clash with the second oscillator and by transposing the first oscillator by one octave it helped the pad have more harmonics. I wanted the pad I was synthesising to have harmonics and not sound like a pure signal.
I used the orgn 3 waveform because this waveform adding more harmonics and brightness to the pad I was synthesising. However, this waveform was making the pad sound too harsh. So what I did was I used a lo pass filter that is located in the centre of the plugin. I used the lo pass filter to cut off some of the harmonics because so that the pad isn't so harsh sounding. I also used a parametric EQ to cut off a tiny bit more harmonics because it was still sounding a bit harsh and tinny. I used a gentle roll off because a gentle roll off is more accurate at filtering out hi end frequencies and I wanted to make the pad sound as subtle possible.
I used a phaser effect that is located on the right hand side of the ES2 and I used this effect so that I can give the pad a stereo image. I wanted the pad to have a stereo image because pads sound wider when they have stereo image and if this pad didn't have a stereo image it would've sounded like mono and I wanted the pad I was synthesising to sound wide and ambient. I then turned the intensity up so that I can hear the phaser working with the pad. If the intensity wasn't up I wouldn't have been able to hear the phaser working and the pad wouldn't have that stereo image. I also adjusted how intensity I wanted to use. Too much intensity would've made the pad sound muddy and this isn't the sound I wanted to hear as I was synthesising the pad.
I used the lo pass filters that are located in the centre of the of the ES2. I used these filters because the pad had a harsh piercing sound and it didn't sound very pleasant to listen to. I adjusted the blend level so that it's central and this way I can use the two filters equally. I didn't want to cut off too much because if I cut off too much I would've taken out the harmonics in the pad. The harmonics is what makes the pad sound bright and stand out. The picture below is the evidence to show what I did with the ES2 to synthesise the pad.
Bass - I used the ES2 plugin to make the bass because the ES2 is really good for transposing octave higher or lower and this benefits more sub or harmonics. Most bass sounds that are synthesised normally use sine waves and sine waves are just pure signals. There is no harmonics in them whatsoever. The ES2 also has distortion control and the distortion control will add harmonics to the bass and the bass will sound fatter. Another good thing about the ES2 is there are three oscillators and three oscillators are good combining waveforms and making a synthesised sound more harmonic and polished.
I used a sine wave on oscillator one, a '00' waveform on oscillator two and the '35' waveform on oscillator three. I then transposed oscillator one and three down an octave. I chose these waveforms because these waveforms were good for giving the bass a strong lo end and harmonics. The transposing of the octaves on oscillators one and three are the key functions that gave the bass that strong lo end. By transposing oscillators one and three the frequencies on those oscillators changed and went lower because the pitch is going lower.
I then used the distortion on the ES2 because the distortion is a brilliant function that makes mostly lo end instruments sound fatter and stronger. I turned the distortion to full because while I was putting distortion it didn't distort horribly and that meant I can keep pushing the distortion until I hear the bass distorting horribly. From there I would turn the distortion down a tiny and this will make the bass sound stronger and fatter. The bass will also have a gentle distortion.
It is really important to be careful with distortion because too much distortion will overpower the bass and will clip horribly. Distortion is stronger than a compressor because the distortion makes the clipping level become a brick wall and by overusing the distortion the instrument or sound will overpower and clip because the distortion is holding too much. I the used the sine level function that's located on the right hand side of the ES2 and what this function does is it makes only the sine waves louder. I used this function because the I wanted the bass to be quiet loud but not too loud. If the bass was too loud it would've caused the output master to clip and distort. So I only put turned the sine level to 50%.
I then used the triangular mixer that's located next to oscillator two and I used my ears to listen to the sound I wanted for making the bass. I used this triangular mixer because this function is really good for combining all three oscillators and listening to for the right sound. I used my ears because I rely on my ears more than looking at the screen. Looking at the screen can deceitful and ears are more trustworthy. From there I used the ADSR on envelope three. I turned the attack and decay down because I wanted the bass to play straight away rather than wait a long and I turned the decay down because I didn't want the decay to gradually reach the sustain level. What I did instead is I let the attack go straight into the release so that I can hold the bass for as long as possible without the bass reaching it's quietest point.
I then used a small amount of sustain and a short release. I used a small amount of sustain because I wanted to make the bass last for a period of time. If the sustain was too short the bass will have stopped playing after a few seconds. Another reason why I used a small amount of sustain is because if I were to make a bassline using the bass I synthesised the sustain will play a key role in ensuring the bass doesn't stop while I'm playing and recording a bassline. I used a short release because if the bass had a really long releases the bass will take a long time to reach zero and the bass doesn't need a long release because the bass is a dry instrument and the bass doesn't have or need a audio tail.
The last thing I did was use an EQ to take out some of the harmonics in the bass. I did this because the harmonics in the bass sounded very harsh. I used a not too steep lo pass filter to filter out those unwanted frequencies. After doing this the bass sounded much subtler and warmer and there a tiny bit of harmonics are still hearable in the bass. The EQ also helped with making the bass more subby and strong. The picture below is the evidence to show what I did with the ES2 & EQ to synthesise and equalize the bass.
Lead - I used the ES P synth to make the lead because the ES P is a synth that is straight forward to use. What I mean by this is there aren't many controls and most of the controls are easy to configure. On the left hand side you can control the waveforms using faders rather then selecting them from the ES2 plugin. There is only one ADSR envelope on the right hand and it isn't difficult to use because you're only using one envelope. There is also chorus and overdrive functions on the left hand side and these two functions are really good for adding effects and harmonics to the lead and leads wouldn't sound great without effects and harmonics. Another good thing about this synth is there is a frequency function located in the centre of the plugin. This function is good for the lead's frequency range. You can adjust how much frequency you want the lead to have and this is a good way of giving the lead harmonics.
The first thing I did to make the lead was select the waveforms I wanted to use. I used a triangle, sawtooth and sub-oscillator -1 octave slider. I used these waveforms because these by using more than one waveform I'm making the lead sound harmonic and the lead will have a big frequency range. I used more of the triangle waveform because the triangle waveform has some harmonics that made the lead sound electronic but the triangle waveform wasn't enough. This is where the sawtooth waveform comes in. I used a tiny bit of the sawtooth waveform because the sawtooth waveform a has a bigger frequency range and more harmonics than the triangle waveform. By combining these two waveforms together I managed to synthesize a lead that has a harmonic sound and big frequency range.
The last waveform I used is the sub-oscillator -1 octave. I used a tiny bit of this waveform because this waveform changed the sound of the lead an octave lower. I didn't want the lead to sound lower because the frequency range will have change if I put too much of sub-oscillator -1 octave. I only used a tiny amount of it because I did want the frequency range to go down but not by much. I didn't want the lead to lose it's harmonics as well as frequency range because the harmonics is what makes the lead sound bright and polished and with the lead an octave lower all of that bright and polished sound would've disappeared.
After using the waveform I used number on the left hand side of the oscillator. The numbers are show high or low you want the octave to go. 16 is the lowest octave and 4 is the highest octave. I chose 4 because 4 is the highest octave and a higher octave was a really good number for me to use. Number 4 helped me with making the lead's frequency range bigger and a bigger frequency range means the lead sounds more crisp. 8 and 16 would've taken that crispiness away and would've given the lead some lo end and this isn't the sound I wanted to hear as I was synthesising the lead. I wanted this lead to have a high frequency range because a lead with a high frequency range can punch through the mix of a song and can be heard clearly without any clashing.
The next thing I used to make the lead is the frequency function that is located in the centre of the ES P. I used this function because this function allows me to control the width of the frequency range. By using this control I was able to control the width of the frequency range and I used my ears to what sounded subtle and what sounded harsh. I relied on my ears more than my eyes because looking at the screen can confuse what your ears are listening to and the ears are reliable and trustworthy at what sounds good and bad.
I used a lot of chorus and tiny bit of overdrive in the lead because the chorus effect gives the lead a stereo image and I wanted the lead sound like stereo because stereo will make the lead sound wider and louder whereas in mono the lead would've sounded central and wouldn't fill the empty space. I also used a little bit of overdrive because I wanted the lead to have some sizzle and fizz. I didn't use a lot because using a lot of overdrive will distort the lead and the distortion wouldn't sound pleasant to listen to. So by using a little bit of distortion the lead sounded a bit more different because the overdrive is making the lead sizzle and fizz.
I then used the ADSR envelope to configure how the lead should be played. I used a sharp attack because I wanted the lead to play straight away after hitting a key on the midi keyboard. I didn't want to wait a few seconds for the sound to come in because with a slow attack I wouldn't be able to make a melody with the lead I synthesised. I kept the attack at 50% because as I didn't want the peak to change when the decay goes into the sustain. I wanted the decay to stay at 50% because it will be easier for me to make a melody with the lead without the volume changing. I didn't want the volume of the lead changing constantly because I synthesised this lead to make a melody for the demo track and the lead wouldn't sound like a lead if the volume levels kept changing.
I used very little sustain for the lead because I didn't want the lead to last very long. I didn't want the lead to last ling because the lead I synthesised has a lot of hi end and a lot of hi end isn't pleasant to listen to for a long period of time. If the sustain had been put to full the lead wouldn't be pleasant to listen to because of the harsh and piercing hi end. So this is the reason why I used a short sustain. The last thing I used is the release. I wanted very little release because I wanted the lead to stop playing as soon a I let go of a key. With a long release the lead would've taken a few seconds to stop playing and if the lead was playing a melody the notes would've clashed together. So this is why I used very little release so that the notes don't clash and that the lead stops playing straight after playing a melody or not being held by a key.
The next thing is I put reverb on a bus and I used the bus to control how much reverb the lead should have. I put some reverb on the lead because the lead sounded dry and I wanted the lead to sound wetter. I put some reverb on a bus because putting effects on a bus is easier to control instead of placing the effect on the channel. By using effects on a bus it allows you to control how much of the effect you want and with ease. This is why I put reverb on a bus for the lead, so that I can control how much reverb I wanted for the lead. If I put too much reverb sound of the lead would drown out and would drown out the rest of the instruments and this is why I used very little reverb.
The last thing I did was EQ the lead. I EQ'd the lead because there was unwanted lo end and too much hi end. So I used the cut off bands to get rid of the lo end and some of the hi end. I also reduced the body of the lead because the spikes in the lead were too big and this made the lead sound a little piercing. So I used a parametric band to reduce those spikes and make the lead sound more subtle. EQ'ing the lead made a difference to how the lead sounded without the EQ. The lead sounded more improved with very little hi end mostly because there was too much hi end in the lead and some of the hi end needed to be cut off because it sounded very harsh and by cutting off the unwanted lo end all the lo end instruments can punch through the mix. The picture below is the evidence to show what I used to make the lead and what I did to improve it.
Electronic and Acoustic Drum Kits - For making the electronic and acoustic drum kits I took some acoustic and electronic one shot drum samples from royalty free drum packs and the first thing I did is I opened the exs24 synth, clicked edit and dragged the acoustic samples first. I didn't need to assign the key because the drums are not pitched sounds and they don't need a specific key to play on. I then assigned each drum to have their own individual key so that all of the drums don't play on one key. The drums were labelled appropriately so I know which instrument is which. The next thing I did is save the drum kit so that all the drum samples are copied onto logic and the drum kit itself is saved. I then repeated the same steps for making the electronic drum kit to make the electronic drum kit.
I used the exs24 to make the drum kits because the exs24 has a sample editor that allows you to drag and save your samples and it comes in really handy when making drums for a song. You can also put pitched sounds in the exs24 and the way you would do this is by finding a sound you like such as a pad, using one note and making that note last 8 - 16 bars and then bounce it as a wav and drag the wav onto the sample editor of the exs24 and assigning the note to the correct key. After doing this save it the sampler instrument appropriately with the sample copied onto logic or otherwise you will lose your samples. After doing all this you can make a chord progression out of that sampled pad.
The exs24 has good functions such as pitch, pan, velocity, key range, reverse, one shot, etc. The reverse function is really useful for reversing sampled drums or chord progressions. By clicking reverse and playing a sampled sound or drum the reverse adds a really cool effect and this can be useful for a fill in or drop. The one shot plays the full sample so it doesn't matter how fast you take your finger of the key. It will still play the full sample. The pitch is a really useful function for playing a sample on different keys. The first thing you would do is adjust the key range of the sample. So from C3 - C5 is a good key range for the sample and the next thing you would do is tick off the pitch function and from there you can play a sampled instrument or sound from C3 - C5 and you would notice the pitches changing. These images show what I did to make the electronic and acoustic drum kits.
Multi Sample Instrument - I used bells to make the multi sample instrument and what I did is made a midi recording on fl studio with the bells I chose to use and I played these notes on three octaves. I then bounced the recording as a wav and I put it onto logic. The next thing I did is strip silence the bells. I pressed 'ctrl + x' because that is the short cut for strip silence and I adjusted the threshold only because the other functions such as 'minimum time to accept as silence' 'pre attack-time' 'post release-time' were fine as they were. They didn't need changing. After strip silencing the the wav file I created 11 new audio channels and I put each strip silenced bell in the channels and I soloed each one of them and bounced them wav files and while bouncing them as wavs I named each bell with the key they are in. I named the bells this way because when I import the bells onto the exs24's sampler the sampler will automatically assign the key of the audio file by reading the reading the name of the audio file and it's a quicker method rather than assigning the audio files manually.
After bouncing all the stripe silence bells I opened the exs24 on a different channel and opened the sampler. The next thing I did is I dragged all the bounced stripe silenced bells to the sampler and I clicked the first option which is "Auto map" by reading the root key from audio file and this option assigned all the wav files to their designated key. I then saved the multi sample instrument with all the samples copied onto logic so that I won't lose the multi sample instrument and samples that are in the sample editor. There were gaps in between the assigned notes and I didn't want the gaps to be there. So what I did is I extended each note to fill in those gaps and I made the multi sample instrument 3 octaves so that I can play more notes on different pitches. I also made sure the pitch function was ticked off on all the samples or I wouldn't be able to play the notes on the higher keys. The images below show what I did to make the multi sample instrument.
Sweep - I made a sweep effect using the ES P synth. I used the noise oscillator only because sweep fx is just white noise and it plays all the frequencies together. I used a sharp attack, long decay, long sustain and short release. I used a sharp attack because I wanted the sweep effect to come in straight away, I used a long decay so that the decay is at the same volume when it goes into the sustain, I used a long sustain so that the white noise can last for a lengthly period of time and I used short release so that the white noise stops after 1 - 2 seconds. The next thing I did is I used a little bit of overdrive and chorus. I used these two functions because I wanted the sweep effect to have some sizzle and a tiny bit of sound. The chorus provides the sound and without the chorus it will just be white noise. So this is why I used overdrive and chorus, to make the sweep effect sound more improved.
I then used a tiny bit of resonance and automated the frequency function. I used a tiny bit of resonance because I wanted the sweep to sound a little bit like a whistle as the frequency is being automated. The whistle sound from the resonance makes the sweep effect sound more natural and I wanted the sweep effect to sound a bit natural because natural sounds are really useful for making a track sound more real like they have been recorded live rather then in the box. The resonance really helped with how the sweep effect sounded compared to what it sounded like without the resonance. I then automated the frequency because the sweep effect wouldn't sound like a sweep without the automation. Without the automation the sweep will just play one frequency constantly and wouldn't sound like a sweep. So by automating the frequency the sweep effect played all the frequencies from lo to hi and it this sounded like a proper sweep effect. The image below shows what I did to make the sweep effect.
Xylophone - I used the sculpture synth to make the xylophone because the sculpture synth has a material function that is to manipulate the sound you are synthesising. These materials are called nylon, wood, gloss and steel. This function is really useful for combining more than one material because combining more than one material helps layer the sound more and it helps develop a better tone for the sound. The pickup function that is located on the left hand side of the sculpture synth is another useful function this function is used for controlling how much pickup to sound should have. A lot of pick up would make the sound too resonant and notes would clash with each other. So using a little bit of pick would be useful for making the notes not clash together and having an audio tail.
I used wood only to make the xylophone because wood is a really good material for making the xylophone sound woody and plucky. Another reason why I used wood to make the xylophone is because wood isn't a harsh sound for the xylophone. The wood allows the xylophone to sound gentle and subtle. I didn't combine wood with another material because wood is the only sound I was going to use while making the xylophone and wood is a great material for making the xylophone sound real.
I then used the pickup function to control how much pick up the xylophone to have. I didn't want too much pickup because too much pickup makes the sound of the xylophone sound harsh and resonant and too much pickup drowns out the original sound of the xylophone and causes notes to clash. So I adjusted the pickup to ensure that the xylophone doesn't sound dry and has an audio tail. This will make the xylophone's sound punch through in the mix and this won't make the xylophone's sound drown out.
The last thing I did I turned the delay off. I turned the delay off because I didn't want the xylophone to have a effect when I played it. The delay doesn't make the xylophone sound genuine because the delay makes the xylophone bounce on and off and I wanted the xylophone to not have any effects or processing because I wanted the sound of the xylophone to simple but real as well. The xylophone was going to be used to layer the multi sample instrument because the xylophone's sound is fits in well with the multi sample instrument. The xylophone has a deep and bright tone and the xylophone sounds plucky as well. The image below shows what I did to make the xylophone.
Pluck - I used the sculpture synth to make the pluck because the I like the materials in the sculpture synth and I can manipulate how the pluck should sound using the pickup function. I used fifty percent steel and fifty percent nylon and I used these materials because these materials allow the pluck to have a bright, shiny and plucky sound. I didn't want the pluck to sound dull and flat because the pluck wouldn't punch through in the mix. A bright sound for the pluck helped the pluck itself punch through in the mix and can be heard clearly.
The next thing I did is I adjusted the pickup for the pluck. I didn't use too much pickup because too much pickup will make the pluck sound too resonating and will cause notes to clash with each other. Another reason why I didn't use too much pickup is because too much pickup would drown the pluck's sound and some of the other instruments in the track, plus, the pluck wouldn't punch through clearly in the mix. So I used very little pickup and this made the pluck sound less resonating and helped pluck not drown out itself or the other instruments in the track.
The last thing I did is I turned the delay off. I turned the delay off because I didn't want the pluck bouncing around. I didn't want the pluck bouncing around because I wanted the pluck to sound simple without any processing in it. I also wanted to hear a tiny bit of plucking as I hit a key because if the pluck didn't have a plucky sound it wouldn't sound like a pluck and this pluck is going to be used an extra melody for the chorus to give the chorus texture. The image below shows what I did to make the pluck.
Re-Sampling - I re-sampled the bass only because the bass didn't have rumble and it didn't sound fat. The bass did have some processing already and the EQ is the plugin that is doing the processing on the main channel the bass is in. I soloed and put a cycle around the bass and bounced it as a wav and from there I dragged the bass onto a new audio channel. From there I processed it some more and the first thing I used is the overdrive distortion. I used this distortion because I can choose which frequency to distort and I can adjust how much distortion I want. I put 6dB of distortion at 410hz because this area was very weak and is the reason why the bass wasn't fat and rumbly. The last thing I did is use the amp plugin. I used this plugin because this plugin gave the bass some warmth and it also worked with the distortion because the amp is making the sub of the bass more stronger and fatter. I used the blues blaster amp preset on the amp and this made this preset improved the bass a lot.
After processing the bass I soloed and put a cycle on the bass I've processing and I bounced it as a wav. From there I dragged the re-sampled bass onto a new audio channel and I compared the waveforms and there was a big difference with what the bass looked like before and what the bass looked like after. Re-Sampling was the best option for the bass because re-sampling a sound, instrument or melody that has already been processed can help get a much improved and developed sound for the instrument, sound, melody, bassline, etc. These images below show what I did to re-sample the bass.
