A Chorus is a device that makes the signal sound wider more harmonious, It does this by mixing the original signal with a slightly delayed or detuned signal.
It is a very simple concept but it is so effective.
This is how a simple Chorus works and below are the basic controls
Dry/Wet - Blends the original signal in with the chorused effect.
Delay - This sets the delay time between the original signal and the chorused effect signal, a long time makes the chorus more loose and short time's make it more centred and focused.
Intensity - This has 2 settings a Amount in milliseconds and a Rate/ LFO in Hertz, you look to get a sweet spot when using this as a high rate and low amount results in a nice chorus and a low rate and high amount results in thick detuning.
Feedback - This sets the amount of signal fed back into the chorus input.
The pictures below show some modern and classic emulation's plugin of chorus's used in the music industry.
So far we have looked at developing a drum loop, layering two sounds together and resampling them together whilst adding subtle effects and mixdown techniques, we then added notes to these sounds and looked at making chords within the C Major scale. This lead us the nice Rhodes & Piano sound in which we added further processing to and then added a bassline to this, overall the track was in a 16 bar loop, we had our main section along with breakdown but now we need to put these loops into song form.
We will look at the structure I tend to go with when I make music, it involves a intro, a outro, 2 main's sections, build section and a break section. As the drawing below shows here's what I use to arrange my tracks.
As you can see the song is divided into sections Intro (24bars), Build (8bars), Main (24bars), Break (16bars), Main2 (24bars), Break to outro (8bars) & Outro (24bars), This gives the song a nice flow to it & keeps the listener interested for just the right amount of time (along with added techniques & FX) before there brain switches off from listening. We will start by adding all of our clips to each section of the song arrangement, and work the sections into each other to get a nice flowing song together.
These are the clips, I have added a few more varieties to the selection. Firstly add all the clips to the arrangement view, you can do this by clicking or dragging multiple clips over to the arrangement view by the TAB key.
When in the arrangement view try splitting up the track into section's, this makes it easier and gives you a guideline on what to work to. You can bypass my example and create your own structure as different genres do ted to have different structures and its always good to try new thing out, you do this by right clicking on the timeline and click on add locator, right click on the locator and click on rename you can now rename the locator as you see fit.
I have arranged the track and added a few more elements mainly to the drum pattern, which is a ride cymbal on bars 49-57 & 89 - 97. I have attached the files HERE for your use, these are the MIDI notes for track so you don't have to worry about altering them to suit or to see how I have done mine.
As you can see below the full arrangement I have used with a couple of added tracks which you will find in the download.
As in this example there are builds, fills, short breaks in order to keep the listener interested. You could try different arrangement techniques such as arranging the layout differently, or adding more variety into the drum loops. We will continue to work on this until we feel its right.
This brings us to our next section Automation, Automation is a process where a procedure is carried out without use of a human, a example of this could be a low pass filter sweeping up through 32 bars to gently bring in our piano. or we could use a Utility to bring in the volume from inf dB to 0dB over 16/32 bars or what ever we want.
To access the Automation on LIVE 10 you must deactivated the keyboard mode by pressing M, and then press A you will see a new view open up whilst in the arrange view (as shown below).
The top picture shows the automation window closed and the picture below shows the automation window open.
In the screenshot below I have shown the basic automation lines, these automation lines are designated to certain effects parameters such at the frequency of a filter.
I will show you the process for doing this below the screenshot.
Firstly we need to decide what effect we are going to use and on what track, I think the having a reverb build on the drum track will give our drop more inpact. First load a reverb to the Drum group and select the automation window by pressing A & make sure you have the keyboard key deselected M.
Next we want to pick the part we would like to automate over time, press once on the dry/wet knob and you will get the parameter in the Device Chooser & Automation control chooser under the group.
When you select a parameter such as the dry/wet under the box's in the group tab select Reverb & Dry/Wet, with this being done press the little + button, this will let you add another parameter on a new line.
With these two now in place we can start to automate, you will see a dotted redline, this shows that value of what you have set it to on the device, to change this simply double click on the dotted red line. Try the following below for a nice snare build up.
You may notice that some of the automation lines are curved, this can be done by holding down ALT/ OPTION & drag and curve your automation to suit.
Have a mess around with automating different parameters or use another third party plugin as shown in the previous blog Plugins & DAW's, overall its a personal thing and do what you think is best for the track.
We can take the automation even further in Ableton and create a effect's rack, this would help us with drawing in 1 automation line via a macro, where we assign numerous parameter and effects to, We will cover this in another blog as for now we are looking at just getting a simple arrangement down, We will keep building on our track and use more techniques to develop it further in the coming blogs.
A De-esser is mainly used on vocals , It reduces sibilant frequencies via volume, sibilants such as S's & T's are most common.
The De-Esser works like a compressor or a dynamic EQ, only it zones in on the selected frequencies, usually the frequencies are between 4-10khz but some plugins may give you more area to work in as shown below in the UAD version its 2-16khz.
With De-Essers usually being used on vocals there is nothing stopping you from using them on other instruments such as Hi Hats, It can be used on the mastering chain to control unwanted high frequencies.
Frequency - Allows you to select the frequency that you wish to use for the sibilants, between 2khz - 16khz on the above version.
Threshold - Activate's the De-Esser above the threshold set value, the UAD version above has a set ratio of 7:1 and a range between -40dB to 0dB.
Solo - Allows you to solo what is being affected by the de-esser, can also be used to zoom in on the frequencies whilst turning the frequency knob from 2khz to 16 kHz.
Speed - This is a set attack / release time, a bit like the compressor/ limiter. The version above is
Fast: Attack = 0.5ms, Release = 30ms
Slow: Attack = 2.0ms, Release = 120ms
A limiter is like compression only that it stops the signal volume going over a set level. It usually has a really fast attack time to stop transients from getting through, the limiter therefore can be used to squash signals closer and closer to 0dB without clipping, this can make a mix sound louder and this is done by reducing the dynamic range of the track. It is usually used at the end of the signal chain on mastering to bring the levels up to a level where the engineer, musician, artist or producer feels like it sounds right.
Limiters usually come with a fixed ratio of inf : 1 and with a attack time set to 0.00ms but as more limiters have been released more are getting more flexible in the option you can use to suit your mix, (as shown in the picture below)
Threshold - A level in dB at where the limiter is activated, anything above the threshold value will be flattened/ shaped to the same level.
Attack - How quickly the limiter reacts to when the signal is above the threshold level is set, or in the oxford limiters case, The ATTACK fader allows the attack time to be increased to achieve an improvement in the sonic qualities of the peak reduction process, by allowing peak transient events to escape hard gain reduction. Since the plug-in has internal headroom, these overshoot peaks are retained and are not clipped.
Release - This is how quickly the limiter reacts to incoming signals in volume, as the release settings on a compressor, there will be a sweet spot between too fast and too slow.
Output Trim - output trim will reduce the overall volume after it has been hit by the limiter, this is good for uploading music to Soundcloud, YouTube etc where the uploading process reduces the overall volume from 0db level.
They are other parameters within the limiter as these above are the main values to get you up and running, further more to this you can learn more about the oxford limiter by visiting the UAD website.
Compression at first too me a while to get your head around but the more I dove into it and the more I understood what the controls did/do and how it is shaped my music I understood how important compressors are. Originally Compressors/ Limiters was used to protect over modulation, this could occur when a signal that was being encoded onto the radio signal, A compressor automatically reduced the volume level when the signal went over a threshold, you can still hear this today on the radio when a presenter shouts or there is a loud noise, the compressor catches the volume over the threshold and reduces it regarding to what setting its has been set too, and then gradually or sharply comes back up to normal levels.
Below shows the main controls for Compression/ Limiters and a brief explanation on what they do.
CONTROL'S OF A COMPRESSOR/ LIMITER
Ultimately it's personal to how you use compressors, giving a overall reducing by 2-4 dB with a 2:1 ratio gives a gentle glue to a overall mix whilst a 15dB reduction and 10:1 ratio will give a slammed feel and bring out the tails in your mix, Its a personal thing.
Compressors control the dynamic range of a signal, changing the relationship between the loudest part and the quietest, it is a very useful and essential tool when used right, it helps create clear and punchier mixes and can also be used on many other individual parts including drums, vocals, synths, bass, aux, mastering/ full mixes etc.
A compressor's main function is to reduce the dynamic range of a signal by reducing the volume of its loudest point/ transient.
Compressors are really effective whilst grouping sounds together, a example would be on a return buss/ Aux buss, they help glue parts together into a cohesive block, for example a drum buss, taking the loudest peaks and reducing them whilst giving the quietest sounds and making them louder, this works well to get the tails of snares, hats etc.
One compressor doesn't fit all, this is so true when looking at compressors, there are not all made to equal each other as some compressors use 2 compressors in series to achieve a unique blend of compression as the Shadow Hills Master Comp shows below
The Shadow Hill Compressor has 2 compressors in series with each other, 1st is a gentle optical section with a set ratio, attack and release (Optical) & then a more aggressive and precise VCA (voltage controlled amplifiers) section (Discrete) each section can be bypass, you can combine these 3 ways, The output allows you to choose from 3 distinctive transformers Nickle (clean), Iron (coloured) & Steel (dirty), Sidechain give us a high-pass filter letting anything below 90hz through.
Every compressor has its own signature, configuration or algorithm which sets them all apart. Software emulations like the UAD-2 version compared to the hardware version is near enough exact.
The pictures below show iconic classic compressors that have shaped our music every since the 60's.
Each one of the above is a emulation of a classic hardware compressor, there are many companies who have emulated these & done a really good job in doing so, but again non sound the same as each other very similar but not the same. If you would like to know more about the setting of the compressor click HERE to be taken to the effects page where you can learn more and about other effects.
EQ is short for Equalisation, It is used to boost or reduce certain frequencies in a sound. Traditional analogue mixing desks have 3 bands a low, mid and high, this allows the mixing engineer to mix the low, mid and high bands independently to transform the sound into what the engineer want's.
There are numerous types of EQ available these include shelving eq, graphic eq, parametric eq, linear phase and filters such as low pass, high pass, band pass & cut, dynamic eq & more, all eq's do a slightly different job & it depends on what your needs are.
This is where you set the point of what frequency you would like to alter, cut or boost.
Cut / Boost/ Gain
Cut or Boost the frequency that you have chosen, you may need to adjust the gain afterwards due to either boosting or cutting the frequencies.
Parametric EQ's width (Q) changes the overall width of that which frequencies have been selected. The bigger the value of Q the narrower the area is.
This dictates how steep the slope drops away in dB, a 18dB slope means that 18dB worth of cut or gain will take place over a single octave.
Flanger was produced by running two identical recordings at the same time and by slowing one down to create a small delay between the two, the short delay produces extra harmonic content which is where you get your classic Flanger fx from.
Flanger has a very short delay time between 0.10ms to around 11ms & also incorporate a feedback control.
Rate/ Amount /Speed - How quick the LFO (Low Frequency Oscillator) sweeps through the spectrum & extends the LFO's influence via the amount.
Feedback - Increases the intensity of the effect by focusing on the resonant frequency.
Hi pass - Filters out any frequencies below the frequency that is selected.
Width - 0% the waveform is identical to the right and left channels at 100% the waveform is in full stereo.
THIRD PARTY DEVICES, PLUGINS & DAW's
This week we will briefly look at what plugin's are and how you can use them in your DAW.
A DAW (digital audio workstation) is a piece of software or electronic device used for recording, editing and producing audio files.
Most modern DAWs have a central interface that allows the user to alter and mix multiple recordings and tracks into a final produced piece.
There are many different DAW's available, some free and some that come with a cost, it down to personal preference and how your work flow is regarding to what DAW you use. Here is a list of most popular DAWS.
THIRD PARTY DEVICES - PLUGINS
Plugins are made by developers which allow us to add modules and effects, in our case into our DAW, these additional devices and plugins can add more depth to our productions.
Effects & Instruments are mainly used as plugins, this allows us to process things differently with the different plugins that we want to use. There are many different companies or developers that all use similar but different algorithms in the coding of there plugins and this is what sets them all apart, nothing sounds the same which is a good thing and this is how a artist or soloist finds what works for them.
Instruments are used for synthesis e.g Moog Mini V, the Effects are used to emulate real-world hardware e.g Pultec passive EQ, Shadowhill master compressor, lexicon 224 digital reverb etc.
Plugins can be used the same way as their hardware counterparts, they are more flexible & cost less, granted you don't get the full replication of the hardware version but as technology is growing and growing the differences between hardware and software is getting closer and closer.
Plugins come I many different formats, these include, VST, VST3, RTAS, DXI, AAX & AU, If you are looking to start building your plugin collection up make sure you know what is compatible for your system & DAW before you purchase, most plugins installation process has a section where you can select from the 4 main formats, VST, RTAS, AAX & AU.
Audio plugins are great addition to producers, you don't have to spend a lot of money to get great emulations of classic hardware, you can use multiple plugins on different tracks, this compared to hardware where you would have to buy multiple units, i.e stereo, they give us more options to use to make music, using the plugins correctly can give our productions that overall polish.
Main sites where you can check overall plugins, latest deals and releases can be found below.
If your looking for a specific company/ developers then the main developers are listed below
There are many, many more plugins available, but the list above will point you in the right direction and where you can learn more about plugins.
Next week we will be looking at our track again and trying to utilise some of these plugins.
For more information about plugins and what the parameters are used for, please visit the effects section in the blog menu.
Layering & Creative Techniques
Welcome to the third instalment learning how to layer & use creative techniques in Ableton Live, in the past two week's we have covered a lot of basic and fun fundamental's, If you have missed any of the written tutorials then please start at the first one Making A Beat, then continue to Chord progression & Basic music theory.
This week we will be looking at another multitude of skills and techniques you can use & see what it can do for us. We will continue from last week and we will be building upon what we have learnt.
We will cover the following in this blog.
LAYERING CHORD PROGRESSION'S
We start by understanding what layering is, lets say we have a bass sound, it's sound is obviously very bassie, we like the bass sound but we think its missing something so we want add another sound to it that will compliment the sound, to show this as a example we will use our chord progression from last week, make a few adjustments and then see what they sound like together.
Open up your previous work from last week and add 2 x MIDI tracks into the Clip/Device Drop area.
Next load the Grand Piano onto one midi track and the Old School Roads onto the other, you can search for this by pressing CMD + F (MAC) or CTRL + F (PC).
Once you have done this click on a empty midi slot, we can now place in notes from what we worked out in last weeks blog, our scale is C major, a nice happy feel to our track.
Enter the following notes into the midi clip on the Grand piano as shown below (left) and enter the other notes on the Old School Roads also shown below(right)
Adding reverb, a touch of delay & compressing the two sounds will make it sound together and in balance. We will first group these two together by clicking on both of the midi tracks and by pressing CMD/CTRL + G, once this is done you should see that the two output from both channels are routed automatically into our new group channel, this makes it easier to compress both sounds and add other effects if needed as a whole, in turn saving cpu power.
We will now add some processing onto our two layered sounds, firstly the Grand Piano (left) & next the old school roads (right) with the overall processing on the group channel below the two.
There is a lot going on in these pictures, The grand piano has a subtle ping pong delay, whilst the old schools roads has a nice chorus adding thickness to it. Overall we have processed the two by adding a nice reverb to make sure the two sounds sound as though there in the same room, Whiles adding the utility and giving it some more width, we then added the saturator to give it a lot more presence we have only changed the curve type of the saturator and you can straight away notice there is a lot more volume and presence, we then added a redux, this reduces the overall quality of the sound via bit rate as you can hear the crackling at the end of the sample, then we overall compressed the two together giving balance & clarity to our layered sounds.
SAMPLING OUR SOUND
Sampling has been the forefront of many brilliant tracks over the years, there is a topic within itself by sampling but for this example we will look how to sample our own sound and use it in our track. Firstly we need to record the overall sound and after our processing this is called post processing, we create a new audio track and set the input to what our group is called in this case its 4-Group, Arm the audio track by pressing the record button at the bottom of the channel, you will notice it turns red & this mean that the track is armed to set to record. Once you have done this press a empty record button within a slot on the audio channel but before you do make sure the two midi tracks are not in LOOP mode, then when your ready click play on the two midi channels, this will record our sound (shown below).
After we have captured our processed sound we can drag the file named 7-Audio-1 into a new midi channel. to do this make a new midi channel and drag and drop the 7-Audio-1 file onto the sample tab. Once you've done that continue to alter the parameters as shown in the 2 picture's at the bottom.
The WARP button is optional I have left it on with the algorithm set to pro, formants & envelopes are also set to minimum and maximum. Try turning the WARP mode on and off whilst having a listen to the sample.
Once you have done this add the following notes into the midi note editor or make you own pattern. (You can then rename the Track of your choosing or name it Piano-Samp), I have also added another noted edit this is for the into of the track, if you notice the velocities of the intro you will see that they are all set to full, this adds digital distortion to the sample by overdriving the sample we turn down the mixer on the main channel by -7.0dB.
Below I have added a pitch bend to show you how to pitch bend inside of a clip using automation, you do this by clicking on the Show/Hide Envelope box in the bottom left hand corner, this will open the envelope box, you will see two box's on top of each other, the top box is what device you want to choose, and the box at the bottom is the control chooser. for the top box select MIDI Ctrl and for the bottom box choose Pitch Bend. You can then make a automated pattern by clicking on the red line as many times you want, you will notice a red dot appears, or you can press B and this will give you a pencil tool on which you can draw in automation to the grid size your in, you can also change the grid size by right clicking on the MIDI note editor and clicking on a certain grid size you would like. For this example I have just added a small curve just before & at the start of bar 7. You can change automation lines from straight lines via pressing ALT (PC & Mac) on the keyboard. This is just a example of how you can utilise pitch bend or other elements depending on what you want within the envelope editor.
ADDING THE BASSLINE
For the Bassline I will add a simple pattern that compliments our piano-samp track, this can be done by playing it in or by copying the pattern used for our piano-samp onto a new MIDI Track, add the Tight Full Bass into this track and press play.
You notice that the track doesn't have any real movement, its very static so we can alter this by making a new pattern or by adding to our existing pattern, add the following notes as shown below & add Notator 16C Swing to our pattern.
With the new notes altered and added, we notice a slight improvement to the bassline, we can make it better again by if we maybe a reverb, delay or both, then un-sync the delay from the grid this will fill in the gaps and add a nice loose feel to our bassline. We will use our BPM to Tempo chart where you can get for FREE when you sign up HERE, and use this to learn what value in milli-seconds we need in order to give our delay a in-synced vibe.
There are many ways to do this but the best way in our case will be via a return track, A return track generally host effects and not clips, multiple channel tracks can be fed into the return track via the sends on the mixer channel which we will show next.
First add a return track by right clicking on the Clip/ Device drop area and select return track. you will notice a new track has appeared next to your master channel, this return track will have a letter in the title starting at A, B, C etc, you will also notice in the track's channel there are rotator knob labelled up with the same letters, these letters are assigned to the corresponding return tracks e.g A goes to A, and with the rotator knob you can adjust how much volume goes through on to the return track from -inf dB to 0.0dB, note if your track volume is at -2db and you have the return track knob all the way up to 0.0dB then the maximum volume you will get on the return track will be the same as the channel's volume, which will be -2dB.
Once you have done this add the simple delay to the return track and click on the two button that say SYNC, you will notice the buttons and sliders are now represented in ms or milliseconds, We want to figure out in ms what value it would be for our track, we will use the BPM to Tempo chart to do this by looking at our chart below, as we want a 8th note delay we will look down the 1/8 note column and we search for the BPM our track is in, which is 120bpm and where the two meet that is our value, which is 250ms, this value shows a synced value very rigid to the grid, if we alter the value just ever so slightly we can make our delay stand out in the track and overall improving it.
Now we have our value we can use this in our mix, we want to subtly de sync the value so in the two time box's in the delay setting's the top box wants to be -1ms (249ms) and the bottom box wants to be +1ms (251ms) as shown below, We then want to set the DRY/WET to 100% as we want the full signal from our track being processed by the delay. Once this is done we can then look at adding SIDECHAIN via the compressor and look to EQ using the EQ eight, you can view a video showing what SIDECHAIN does and how we can use it within our mix's. Copy the values from the picture below and press play.
Below shows the Channel return track value set to full and the return track volume value. As you can hear this is a very subtly effect but it is very effective to fill in the gaps from our bassline, along with playing it against the piano track the bass compliments it really effectively. We can add the piano track to this by turning the sends knob on the piano track to the desired amount, in our case full.
COMPRESSION & FURTHER SIDECHAINING & LEVELS
We now want to get the levels of our track right so the Bass & Piano sits nicely together and with the drums, the bass want's to be at -12.9 & the Piano wants to sit at -2.6.
Next we want to group the Bass & Piano tracks together, we do this by selecting the Bass & Piano track and pressing CTRL/CMD + G, with these now grouped together we can add effects to both of the tracks at one time, this helps with keeping the load of our cpu down.
On the group track add a glue compressor and a compressor, we want to use the glue compressor to control the overall and the compressor for further sidechaining.
Put in the values & levels as shown below along with sidechaining the compressor to the kick.
As you can see if you press play the bass sits well along with controlling the piano sound and letting the drums cut through the mix, this technique is used not just for the pump effect but it can be used against other elements that are clashing with each other, as we have demonstrated with using the delay as a return track.
This week we have covered many main points within making your music creatively and also technically, using these techniques can really make a mix stand out so extra care when mixing your tracks as too much of something could effect something else within the mix, you've got to trust your ears and do what you think is right.