When it comes to tracking any Hi-Gain guitar tones there's one obvious thing that causes most audio engineers a headache and that is unwanted hiss & hum coming from the amp/speaker cabinet. This can even be the case with Guitar Di's as the noise floor can be raised by the pre-amp or audio interface it's plugged into.
I've seen some very elaborate methods employed in the attempt to eradicate it, some good, some bad, but there's one simple method that is usually overlooked.... inserting an Expander.
Now for Studio One users this proves ever so simple to implement as S1 comes with a free Expander plugin and for most, on a tight budget, you can't get better than having something offered for free. Firstly lets try and explain what an Expander does.
Expanders increase the dynamic range of a signal such that low-level signals are attenuated (Volume decreased) while the louder portions are neither attenuated nor amplified. This is effectively the opposite of compression. Use an Expander to decrease the levels of unwanted noise or bleed from other sources or to restore dynamic range to a compressed signal.
Now to keep things really simple and uncomplicated S1's Expander plugin comes with a built in preset called 'Noise Removal'. That, like all presets, is a great place to start. Watch the video below as I demonstrate how best to use it to achieve an optimum signal to noise ratio.
The secret is eradicating the noise without eradicating the guitar signal.
For those who want a bit more extensive read on what all the controls on S1's Expander plugin do here's the list below taken direct from the wonderful Free Studio One User Manual.
The following parameters are available for the Expander:
Threshold Adjusts the maximum amplitude at which processing occurs. Variable from -60 dB to 0 dB.
Range Adjust the maximum amount of attenuation applied to the signal. Variable from -72 dB to 0 dB.
Ratio Adjust the ratio of the Expander. Variable from 1:1 to 1:20.
Look Ahead Click to engage/disengage 2 ms Look Ahead function.
Attack Adjusts attack time for dynamics processing, reaction speed to falling signal. Variable from 0.1 ms to 500 ms.
Release Adjusts release time for dynamics processing, reaction speed to rising signal. Variable from 50 ms to 2 s.
Sidechain Engage by clicking the [Sidechain] button at the top of the effect window to allow other sources control the Expander.
Sidechain Channel Display Displays the current sidechain channel source.
Listen Filter Click to listen to the filtered control signal of the internal sidechain. This helps find a specific target frequency for the control signal when de-essing, transient damping, etc.
Lowcut and Highcut Frequency selection for internal sidechain filters. LC variable from Off to 20 Hz to 16 kHz; HC variable from 20 Hz to 16 kHz to Off.
Swap Click to swap the frequencies used for LC and HC.
The iconic 1176 compressor is a well renown stable piece of audio equipment that is found throughout most professional studios. It was the one go to compressor that could be used on just about anything from bass to vocals due to its fast attack time.
What really got people excited about the 1176 was not just how it compressed things or the 'All four buttons mode' but how it injected character into any audio passed through it. Some engineers engaged it's circuit without using any of it's compression/limiting abilities just to have the character, of it's all transistor circuit, effect the audio travelling through it.
The 1176 is said to brighten the audio going through it almost acting like an Eq move but most definitely gives audio clarity and improvement in the top end that most engineers find a welcome addition to their workflow.
However you feel about it there's no getting away from the fact that this is a stable piece of equipment and completely deserving of it's iconic status. The simplicity of operation, compression, limiting that it has makes it a highly desirable piece of outboard gear.
1176 Plugin
For those of us who haven't got thousands of pounds/dollars to splash out on an iconic 1176 hardware compressor we are left with one other option and that is a plugin software emulation. There have been many various attempts to capture the essence of what the real world counterpart does to audio but they have always been a little off target.
Whether it be the response or the actual character, that hasn't been just right, plugin manufacturers are still pushing the boundaries and with more powerful CPU's being fitted to home computers are making gains toward being 100% accurate on simulating these hardware units.
There have been many versions from companies like Waves, UAD (The original hardware manufacturer), Softube, IK T-Racks and others but while they have come close there was always that little something missing. Now step forward the new contender to the throne PSP Audio FETpressor.
PSP FETpressor
Is a FET feedback type compressor plug-in. It’s designed to provide accurate, almost instant compression with lots of character. Great for vocals, guitars, bass, drums, and whenever a classic 1970s-style FET compressor is needed. PSP FETpressor contains everything you’d expect from a FET compressor: an internal side chain high pass filter to control amount of low frequency pumping, smoothly adjustable compression ratio, ability to select which channel to process, and a link switch. This compressor also contains a dry signal blend for parallel compression. Additionally PSP FETpressor contains makeup amplifier and output transformer emulations to add subtle character even when set to 1:1 ratio.
Now I'm not one to endorse or advertise any software save that of Studio One or be a victim of the 'This plugin will change your mix's from amateur to pro', but after trying this out (Demo available here: PSPFetpressor ) I have to say this is the closest thing I've ever heard to a real 1176 hardware unit. It definitely has the character and the compression/limiter down to a 'T' and has become my go to compressor for everything even on the Masterbus. It can be a monster runaway train that pummels you to the ground or as subtle as a feather landing on your head and offers a wider range of options than an original 1176. But the one thing that sets it apart for me from the rest is the character it imparts. Just like the real unit and that is something extra special.
Conclusion
So all being said and done don't trust my ears or take my word for it go download the 14 day demo and see if you agree. It's what I've been waiting for a plugin counterpart, to the real world hardware unit, that undoubtedly ticks all the boxes when it comes to feeling like your actually throwing an 1176 onto something when tracking or mixing. So watch the video below as I demonstrate the FETpressor on a few tracks and don't delay, give it a go yourself and see if your not convinced there's never been a more exciting time to being a bedroom/home studio producer who simply hasn't got huge budgets.
When it comes to using reverb a lot of mix engineers will give it a wide birth claiming they just can't get it to work for them and for many that is indeed the case. But despite an ever growing repugnance of reverb in favour of delay it still has it's place in the mix.
So I want to re-address this dislike for reverb and show how it can be a great tool for any mix there's just a few things to understand first as to how to get the best out of it.
What is Reverb?
Think of it this way, in it's basic form it's just and echo. Imagine standing in a cave and shouting "Hello!" You hear all the reflections of the cave walls making your voice sound bigger as it tails away to nothing. Or have you ever stripped a room out in your house for decorating and suddenly you hear the echo and space in the room? That essentially is reverb, it's the reflection of the sound in a room. The less in a room the more reflections, the more in a room (furniture, people) the less reflection as they dampen the sound. So now you have a real world example of what reverb is now you can utilise that knowledge when using reverb in a mix.
The difference between delay and reverb is that reverb is generally real world rooms and spaces.
Types of Reverb
HALL: A hall is a large space either in length or height that has a long decay time and lots of reflections. Classically used on John Bonham's drums where they used the height of the hall and staircase to create the massive drum reverb sound loved by many.
ROOM: A room is a much smaller space that can be dead or reflective, depending upon the material
that the walls, floor, and ceiling are made of or the amount of items in the room. It usually has a short decay time of about 1.5 seconds or less. This is the most useful of all reverbs especially if the band didn't get tracked playing live in the studio all at the same time. It can be of great benefit to recreate that by using a room reverb and sending all the instruments and vocals into the FX channel, adjusting the volume of each send to mimic the live band feeling.
CHAMBER: An acoustic chamber is a dedicated tiled room that many large studios used to build to
create reverb. Famous examples are the ones created by Phil Spector 'Wall of sound' at Gold Star Studios, sadly now gone and others like the one at Capitol designed by Les Paul revered by many mix engineers. Extremely expensive to build hence their scarcity but the bedroom producer can mimic it somewhat on a lesser scale utilising a tiled bathroom for instance. Sound was usually passed through a loud speaker and then a microphone would pick it up again, including the effects of reverb. Although this is still common the cons with this technique is that varying the reverb time can be more difficult. A chamber is usually a smaller physical space than a hall that results in more clarity
PLATE: Introduced as a cheaper alternative to building a chamber and the first artificially created reverb, is literally a 250kg, 4x6ft, hanging plate of sheet metal with transducers attached to it. Early units had one pickup for mono output, later models featured two pickups for stereo use. The reverb time can be adjusted by a damping pad, made from framed acoustic tiles. The closer the damping pad, the shorter the reverb time. However, the pad never touches the plate. Some units also featured a remote control. The most famous of these was the EMT140 made famous by Abbey Road studios.
NON-LINEAR: Again another artificially created reverb and solely available on digital reverb units. The pros of these reverbs, such as the infamous AMS RMX16, is that they could create reverbs that just were not possible in the real world. They also allow for special effects such as the reverb tail being reversed so it builds instead of decaying. This gives them an advantage over their real world counterparts in that they can really produce some otherwise unobtainable special FX's.
So hall, room and chamber are all real world rooms whilst plate and non-linear are, like delay, artificial. When it comes to choosing which reverb to use well there are no rules but in general, room or a chamber on drums, a plate on vocals or guitars, and a hall on strings or keyboards are the most common choices.
It's All About Timing
Slapping a reverb effect on anything without addressing the timing is pure folly. You need to specify the matching time of the reverb to correspond with the beats per minute of your track. This will help them blend in better in the mix and sound more natural.
Working out the timing of a reverb or a delay for that matter can seem daunting but fear not there are lots of lovely people who have offered calculators online such as Nick Favour that take all the hardwork out for you.
For the more inquisitive a manual way of setting up timing is to send the snare to a reverb Aux/Fx channel. Solo the Aux/Fx channel and adjust the decay/release time on the reverb plugin/hardware unit. What you are aiming for is after the initial first hit of the snare, listen for the reverb as it decays and drops in volume. Now lower/raise the decay or release time of your reverb until the tail decay just about stops before the second hit of the snare sounds. Now your good to go.
The Importance Of Pre-Delay
Another feature of reverbs not to be ignored is the Pre-Delay control. Pre-delay means delaying the sound of the reverb slightly after you hear the source signal. So the direct snare sounds first and then shortly after that the reverb. The reason it’s used is so the source signal doesn’t sound washed out in ambience. With a little bit of pre-delay, you’ll hear the source’s attack, then the reverb, so the source signal has more definition. A good starting point is around 20ms maybe up to 40ms somewhere in that region. When setting up, pre-delay, is more important to the overall effect of reverb as it is the one thing that will blend and create a natural effect in the overall mix.
Conclusion
So reverb can be of great benefit to your mix when you gain an understanding of what reverb is, how to crucially set it's timing and how using Eq can shape it to suit. Watch the video below as I endeavour to show some of these ideas and how to implement them.
When Console Shaper was introduced to version 3 no-one really knew how to use it properly and things were a little sketchy even from Presonus. I mean we all got the concept behind it but there seemed so much ambiguity on how to best implement it.
Things then got a little more interesting when we were promised third party manufacturers would be developing more console simulations for Console shaper. Well we waited, and we waited and well nothing appeared. That is until now. Details are still a little sketchy at best but Presonus have wheeled out a serious update and here's their video showing some of the features that will be introduced.
This is what Presonus have to say about it:
The CTC-1 Pro Console Shaper by Presonus is the second plug-in developed for the new Mix Engine FX interface in Studio One and the next evolution of plug-ins from the included Console Shaper. CTC-1 brings analogue console modelling to Studio One with three great-sounding console models and several major enhancements to the Mix Engine FX environment.
Utilising state-of-the-art State Space Modelling technology, CTC-1 captures the character and personality of legendary analog consoles that are otherwise out of reach for most musicians, producers and engineers. Going far beyond virtual channel strips and mastering plug-ins, CTC-1 is capable of changing the way how people mix inside a DAW. Priced around £76.
I play the way I do because it allows me to come up with the sickest sounds possible. That's the point now isn't it? ~ Jeff Beck.
When it comes to guitar playing we can talk about technique, showmanship, guitar manufacturer and gear but all that falls by the wayside if that one all important element is missing... 'Tone'. What would any guitar hero's reputation be if it wasn't for them having the expert ability to 'dial in' that killer tone for the sickest rhythm sound or rip roaring solo. The more elements you have (Amp, Compression, Eq, Modulation, Distortion, Delay, Reverb) the more time it will take to find and the more confusing things can become. It's easy to get lost in the maze of choices available so why not keep it simple and build on that from there.
In the Pro Studio having a 100W Marshall/Fender/Rectifier cranked up to find that sweet spot required, where the amp just starts to break, is no problem having nice isolation rooms so you don't go deaf listening to it, but for the bedroom producer, that's just not an option. There's neighbours, other family members and even Fido and Garfield the cat to not scare the 'by-geezas' out off when you suddenly press record and strike that first power chord. Nowadays manufacturers are aware of this and have went out to capture that specific market with offerings of 1-5W versions of their amplifiers. This allows for a lower volume setting to hit that all important sweet spot on a tube/valve amp without measuring 7.5 on the Richter scale.
Now this is great news for the home recordist with neighbours and no soundproofing but those amps are still quite expensive and give a one amp sound. For most the mainstay of their guitar playing and recording will be through Guitar amp simulators such as Amplitube, Bias, S-Gear etc that can give loads of various amps/cabs/mics/pedals for far less but for many purists they completely miss the target of sounding authentic.
Now you could definitely argue that point out all day long but truth be told, like plugins are to hardware, the gap between a real tube amp mic'd up and a simulation is closing all the time. Watch online the cream of the crop simulator, the Kemper, and be amazed as to how many purists have failed to tell the difference. Again a Kemper is extremely expensive but considering how much you would have spent on all the gear it simulates and well you wouldn't have much change from having spent tens of thousand of dollars/pounds.
Software Guitar Simulations
So if your recording at home in a bedroom/kitchen/basement it's safe to say you don't have huge budgets to splash out on gear and so the only option left open to you is software simulations and that is where I want to aim this at.
Dialling in a tone to any of these simulations is not as simple as just loading up one of their extensive presets, unfortunately it's not that easy. Rather like the real world counterparts you need to dial in your tone and be aware of how the real equipment responds so you can simulate it. Now presets give the wrong idea that all of that has been taken care of for you, but don't be fooled. There are some basic things you can do before hand to get as close as possible to mimicking the real amps & cabinets.
First up - Audio interface. Most two channel interfaces have a dedicated ¼" instrument jack. Now some claim to be Hi-Z this is supposed to mimic a guitar amp socket but don't be afraid to try out the mic/line input instead. The instrument input is clean, untouched, but the pre-amp mic/line input can add some colour and warmth to the signal that really helps in taming those upper harsh digital frequencies that give away the fact you've played through a guitar simulator. After all, in real world conditions, the mic on a guitar cab is always through a mic pre such as a Neve/Api/Console that all colour the sound.
Secondly - If your just choosing the in-software cabinet simulation that comes with your Guitar Sim of choice then your losing out massively on realistic tone. Most believe that ⅔ of a guitar tone comes from the speaker cabinet. So with that in mind wouldn't it be prudent to get as close to that as possible? Fear not that is indeed achievable for the bedroom producer in the form of Cabinet Impulse responses, most notable, those from Rosen Digital, Redwirez and Ownhammer. Go to any of their Websites and you will find they all offer a free sample, so what are you waiting for? Mimic a Mesa boogie 4x12 cabinet for FREE. Now that's what I call low cost. If on the other hand you can afford to splash out a little cash? My recommendation is the Justin York Collection from Ownhammer simply sublime.
Thirdly - Copying the real Eq frequency response of a real cabinet. Now why do you need to do this, isn't that what a Cabinet IR is? Well yes and no but knowing that speaker cabinets produce next to nothing below 75hz and the same in the upper frequencies with anything over 5k and you can see there's a range where that sound needs to stay and resonate in. That can vary greatly if you then add further processing to your guitar track with Eq, Compression and tape emulation. Also note Marshall, Fender and other speaker cabs have a mid frequency dip too around 400hz for the Fender and slightly higher for the Marshall. see Fig below.
So for instance SOUNDING LIKE A GUITAR TUBE AMP there is a very quick, simple step that can be taken to duplicate the sound of a guitar tube power amp with the Presence and Depth set to 0, similar what is offered in other OwnHammer speaker cabinet impulse response libraries. To replicate this sound, following the cabinet IR loader add an EQ with a parametric bell curve set to -3 dB at 400 Hz. Adjust the Q/bandwidth to roughly where the edges of the curve start to make the initial cut around 100 Hz on the low side and 2 kHz on the high side. If necessary, adjust the Q/bandwidth to taste from here to best suit your sound source and tonal preference. Similar to this:-
Here you can see I've dipped out around 400hz adjusting the 'q' so the shoulders of the dip hit around 100hz up to 2k to mimic a Fender Cab. I've also added a low pass filter at 5k with a slight 6db slope, nothing too aggressive, to mimic the fact that most speaker cabs don't produce anything above that range.
Conclusion
Now of course tone and everything surrounding it is extremely subjective. Look on any forum that touches on it and you'll see quite heated debates. We all have an idea of what a killer guitar tone actually is and they are all different. Some love Les Paul's through a Marshall, others a Fender Strat through a Fender 2x12, so who's to say what is 100% right or wrong. But that's not the aim of this example. It's about getting as close as possible in sound to the real thing when using a real tube amp in the urban setup is just not practical. If you can afford a Kemper then great your already there but for most modest setups software is the only option and that's what we're trying to improve on here.
Watch the video below as I show and discuss some of these ideas.
Well it's that time again a new update for Studio One 3 to version number 3.3.2.40899. This one kinda crept up on us all. It's main new feature is integration ready for the new hardware release from Presonus the Fader Port 8.
You can download the update from within S1 on the options page or go to Presonus, log into your account and download it from there. Here's a list of all other fixes and improvements:-
Version 3.3.2 Release Notes (November 29, 2016):
New features and improvements:
• PreSonus FaderPort 8 Integration
• Modulation wheel position is saved with the preset or song
• Removed notes overlay on non-Melodyned events
• Copy/Paste for VST-XML (Revoice Pro)
• Video Engine tweaks
• Plug-in editor does not close now when disabled via editor menu
• Copy audio event to new track also copies all takes/layers
• QWERTY Keyboard Remote Device access via CAPS lock
• Encoder remote control for pan and other controls is more usable now
• [macOS] MIDI port names are more readable now
• Copy automation via copy and paste events
• Plug and play for Windows USB MIDI devices (Reconnect is automatically initiated
when plugging in USB MIDI device.)
• MIDI program change for VST3 plug-ins
• Templates for StudioLive AR mixers
The following issues have been fixed:
• Browser not scanning Sound Sets in subfolders
• Lockup when pushing corresponding number on keyboard for Select Alternative Tool
• Event content is drawn outside event boundaries on tracks & layers
• Studio One will ask to use audio device that's never been connected
• Not possible to insert a send effect to multiple selected channels
• Crash after recording Instrument part
• Crash after Glue Instrument Parts
• Quantize audio events/parts ignored first selected event per track
• Studio One crash in Rewire mode
• Artist picture has color tint when using CMYK JPGs
• [macOS] MTC out of sync with NLA Video Slave 3
• Presence XT does not store active key switch
• Melodyne Install Now link broken
• “Between each Marker” export bug
• Writing Automation with the Range Tool will affect everything during playback
• Instrument editor is stealing focus when open
• Part Automation changes without user action
• Crash on close all
• Wrong pre/post fader icon on main channel
• Crash when closing Studio One while updating
• No "missing samples" dialog after drag & drop of presets to the layer list
• [ARA] No UNDO in Studio One for tempo changes in Melodyne
• Copy single automation node not working
• Z key (American Keyboard) activates Y key on QWERTY Keyboard
• Disabling first insert on instrument channel disables Note FX
• Video Player fixes for Windows and macOS
• Auditioning MIDI files in Browser not working
• Alt + Drop FX to Send panel doesn't respect multi-selection
• Filtering loops by category can take very long
• [Win] Moving plug-in windows between different monitor DPIs broken
When it comes to recording vocals one of the most unwanted artefacts that can protrude from the vocalist or processing on the vocal via the mic, compressor or eq is the dreaded "Sss" "Shh" known as sibilance. Sometimes the sibilance can be so predominant it can take your ears off with the sheer shrill frequencies like nails down a blackboard. A vocal can sound reasonably fine but slap on a compressor to help level things out and then you can find the sibilance is like a jet aeroplane taking off.
Automatic
The way to combat sibilance is to use a de-esser. A unit or plugin that compresses just the 'S'
frequencies that usually fall somewhere between 3 kHz and 10 kHz, depending upon the singer or processing in use. It's like a mini one band compressor.
The most common classic hardware de-esser dbx 902
The hardware units like the classic dbx 902 shown above were limited with only two controls threshold and frequency. The Threshold control sets the level when the de-essing process begins. The Frequency control allows you to fine tune the frequency to exactly where the S’s occur. Now when it comes to plugin De-essers we are presented with far more precise control and they are able to do more complex tasks than their hardware counterparts could ever do.
Some of these other features include;-
Split-Band - Here, the signal is split into two frequency ranges high and low bands, a range that contains the sibilant frequencies, and a range that does not. Allowing for only the 'Sss' in the selected frequency to be compressed and the rest of the signal to go untouched.
Wideband - As it's name suggests lets you target a wider area. This can be useful on instruments rather than specifically used on vocals. For instance for the removal of finger squeak on an acoustic guitar track. However on a vocal it tends to leave the leading edge of the 'S' unalterterd and you'll tend to catch some other non-sibilant words and reduce them too.
Sidechain - A little confusing terminology found on some plugins as it refers to simply the ability to solo out the de-essing and help you listen and target the 'S' sound more effectively.
One other automatic way of de-essing is to use pre-triggering of the de-esser. This can be achieved by making a duplicate copy of the vocal track and move the copy 50ms ahead of the main vocal. Now you insert a de-esser on the original vocal track and key it's side chain to the copy. If your de-essing lacks a side chain facility then try using a multiband compressor that has that function. It will act the same way.
Through this method everytime an 'S' comes through on the copied track it will activate the de-esser on the main vocal track but 50ms earlier than when it will sound on the original track. This allows the de-esser to reach peak gain reduction before the 'S' actually sounds. The benefit of this method is a more natural transparent sound.
Manual
De-essing by manually seeking out those annoying 'S's can be painstakingly long but more rewarding in my view as it sounds more natural. This is best done by soloing the vocal in question and listening for any 'S' 'Shh' sounds that really stand out too much. Once found you have two methods of reducing their volume, much like an automatic de-esser hardware/plugin does.
Remember to first duplicate your vocal track.
Never do this amount of editing on your main vocal track or you could get lost or really mess things up and find you have too many undo's or worse find no way of getting back to your original.
First is by using the clip gain reduction. This is were you would find the troublesome 'S' , cut the audio each side of it and pull down the event/clip gain by -5 to -6db. On playing back it should still be heard but not be so much in your face so to speak.
Second method of manually de-essing is by using automation. This is slightly more easier to do as you don't need to slice or split up the audio. Like the method above locate the 'S' and using volume automation draw in a -5 to -6db volume decrease. This means as the 'S' begins to sound the volume dips lessening its audio impact.
Conclusion
Which ever method you choose one thing is for sure, you want to lessen the impact of any 'S' sounds but not take anything else away. Most automatic de-essers no matter how well setup and tweaked will always reduce some of the frequencies that your not wanting to target. This is where manual de-essing has the upper hand. It's far more specific and precise honing in solely on the annoying 'S's and leaving the adjacent frequencies untouched.
Note it's very easy to over DeEss. If your singer sounds like someone took out his/her front teeth you
are probably overdoing it. If you hear the “DeEssing effect” that too is usually a sign of over
DeEssing. If the audio passage sounds rather natural and free from annoying Sizzle and distorted
Esses then you got it right.
Watch the video below as I demonstrate the pros and cons of automatic and manual de-essing.
“He who sings scares away his woes.” ― Miguel de Cervantes Saavedra
When it comes to mixing there is no other element more important than the vocal. Engineer's will allocate hours, days, however long it takes to sculpt that perfect vocal. They'll throw every thing they got behind it to make it stand on top of the rest of the mix. After all it's the singing that tells the story, evokes the emotion and pulls on the heart strings. It's the vocal that spins out those unforgettable lyrical lines, that seem somehow, more spiritual effecting our thinking, our moods, than even the staunchest atheist would care to admit.
Now for some the thing that draws them to a song maybe the guitar riff, the bass riff etc. but for me personally it has always been the sound of the vocalist. Add to that meaningful lyrics that provoke thought and interest and I'm gone.
So taking all that into consideration the importance of the vocal, usually the final and last thing to be mixed for most engineers, can't be stressed enough. Throw in a thousand different things to be considered from dealing with plosives (P and S sounds close mic'd) to automating levels on the most transient of all instruments, the human voice, and you've got your work well and truly cut out for you. You never want a single word to fall by the wayside or not be heard or have it buried beneath the mix. It has to somehow float above everything in it's own space and getting that right requires years of experience. You can't just slap on a compressor, add a smiley face eq and some reverb and think you're done, no, this goes way beyond all of that.
Watch below as I offer some popular methods of creating that perfect vocal, and no I don't mean moving faders like a man possessed and turning on autotune, that unfortunately has it's place but when you are presented with a true super talented vocalist, who can actually sing, it's your job to make sure everyone gets to hear and enjoy the exact same experience that your own ears have just been treated to.
Never a truer statement has been made when it comes to the stock plugin known as the Mixtool. This little somewhat insignificant plugin definitely punches above it's weight. It has several functions and would sit quite happily amongst the company of Pro Tools Trim tool but it improves on that by giving a few more options.
Stereo track version.
Mixtool provides common track utilities, including independent left- and right-channel inversion, left- and right-channel swap, and MS transformation of stereo signals. Use Mixtool when inverting channels to correct for phase cancellation and correlation issues, as well as to provide MS transformation to decode signals recorded with Mid-Side stereo configurations.
The following parameters are available for Mixtool when used on a stereo Track:
Gain Set overall output Gain (Volume).
Block DC Offset Re-centers the incoming waveform, to remove any DC Offset in the audio signal.
Invert Left Click to invert the phase of the left playback channel for a stereo Track.
Invert Right Click to invert the phase of the right playback channel for a stereo Track.
Swap Channels Click to swap left and right Mixtool input channels. Stereo Tracks only.
MS Transform Click to perform a Mid-Side transform on the Mixtool input channels. Stereo Tracks only. Generally used to decode MS-recorded signals or to create MS signals for stereo image processing.
When used on a mono Track, the Mixtool plug-in has one control to invert the phase of the signal.
Mono Track Version.
Watch the short video below as I show a few examples of this excellent little utility.
The guitar has a kind of grit and excitement possessed by nothing else. Brian May
When it comes to Guitarists and fitting things in a mix there's one sure thing you can rely on that every guitarist will say? - "Let me just add one more guitar"
Despite there are only so many frequencies to share around with the other band members I've never met a guitarist yet who was into using minimal guitar parts. They will always be the one who will present the biggest challenge in trying to convince that they don't need 20 guitars playing on the left and right to create a wall of sound. In fact in reality it's quite the opposite.
The trick here is good man management. You don't want to seem like your giving more attention to one member of the band over another but winning over the guitar player/s can seriously cut down on time wasted in the studio arguing about how many guitar parts a song needs. Decisions like equipment (guitar, amp, pedals) and tones (committing to a sound) should all be sorted out before even entering the studio. Sure it's fine if you suddenly and unexpectedly come across some random stroke of genius, always be recording, but within reason. The greater percentage of what's going to go down in the studio should have been ironed out in rehearsals, discovering whats working and whats not. But there are some constants that everyone should adopt into there playing.
Layering Guitars
The whole emphasis of layering up guitars can best be shown by one of the greatest session guitarists that ever graced a six string ax in his video below "Layering Guitar Parts":
So many examples and tips from Tim he best describes the methodology required to create awesome guitar parts that can be utilised in any genre. Less is more especially when wanting to create big giant and wide guitar parts. Notice how Tim plays early tracks with no reverb allowing him to have it only added to solo parts. Also instead of playing roughly the same guitar part to the left and right he alternates with different chord shapes (add a capo for example), using single low end and high end harmonic guitar parts so the sum of all parts add together to create something really special.
Panning Guitars
When it comes to panning everyone automatically defers to thinking first two guitars panned hard left & right followed by another two panned 75 left & right. But if you go 100% left and right from the start of the track then there's no way of adding in further panning options to give, for instance on a chorus, that added width giving the track some dynamic change and stepping things up a level.
Of course there are no set standards but consider having your main two guitar parts panned 40-60 left & right for the verse and then go full 100 left & right on the chorus for greater effect. Automation can be your best friend here.
Guitar Tone
Other things to consider are having distinct tones on the left & right especially on similar guitar parts. Try to mix things up by using different guitars (single coil, humbucker) different cabinets, mic position, using a capo, Eq's anything to make the left and right tonally different. You can test how well two guitars panned left & right work together by dropping your mix temporarily into mono. Here you'll notice if you can still hear two distinct parts, if one is drowning out the other or if one completely disappears. If that's the case select a different tone by utilising one of the above methods.
Watch the video below as I show and discuss adapting some of these methods into a mix.
None of us wanted to be the bass player. In our minds he was the fat guy who always played at the back. Paul McCartney
When it comes to being the bass player for many of us it's the least coolest member in the band but take the bass part out of any song and well.... there's a huge vacuum of sound and dynamics. When we consider some of the greatest bands, a much overlooked fact is how amazing the rhythm section was, and chemistry between the Drummer and the Bassist, that drove the whole band along. Such rhythm duo's as Jones/Bonham, McVie/Fleetwood, Redding/Mitchel, Bruce/Baker so on and so forth that propelled these bands to the giddy heights of super-stardom. So we must never underestimate the role or the part they played in musical history and neither must we ignore these parts when it comes to mixing.
Laying A Foundation
There are no two elements that fight against each other more in a mix than the Kick drum and the Bass guitar. Getting these two to live in harmony is crucial to building the rest of your mix. Get it wrong and the whole track can be ruined before you've even started. Now of course these things are personal and subjective but most mixers start by getting the drums and bass to sounding just right before moving on to the rest of the instruments.
Tackling the low end can prove more difficult than first thought but there are some basic methods commonly used to getting drums (especially the Kick drum) and Bass to live in sweet harmony wth one another.
Rule No.1# Don't forget to change your Bass strings!
Watch the short video below as I show some of these methods that are used routinely by many mix engineers who have found them to be tried and tested methods, hey if it ain't broke why try to fix it, am I right?
Sometimes when your dealing with clients and they've sent you the stems of their new found creation there can be issues that arise beyond your control. When these things happen you have to decide whether to carry on trying to fix things, before you can progress, or to just call it a day and being ever so polite return to sender.
Remembering what your specific job role helps greatly in making a solid decision. Your job is to mix! That's what you've been contracted for and whilst having the ability to bend a little and be somewhat flexible, if someone has failed in their job before you then surely it's their problem and not yours, am I right?
In the example below the track comes pre-loaded with some serious issues. Most notably wrong mic selection, overloaded mic's and heavy distortions throughout. But being as you never want to admit defeat when facing any adversity you don your detective hat and spend an hour or two trying to work out what has went wrong at the tracking and/or pre-mixing stage. But sometimes you need to step back and with a clear head admit defeat and move on.
Watch the video below as I demonstrate some issues with this particular track.
There's lots of examples of how to do parallel compression and the benefits it can add to a mix but there's one technique that supersedes them all, the original and the best, termed by mix engineer Bobby Owsinski, and that is 'The New York City Compression' technique.
NYC Compression first termed by Bob Owsinski
Now at first look you might think what makes this any different to ordinary parallel compression, isn't it just sending or copying to a another track/bus/fx channel your original track, add a compressor, crush it and blend it in?
Well the answer to that is in part yes, but then there's more to it than that a few extra steps that set it apart from the rest and give it an edge over other methods.
Watch the short video below as I demonstrate how to implement it and how it can be used to add excitement and extra dynamics, especially to the rhythm section (drums & bass) of any mix.
When it comes to mixing drums, specifically multi-mic'd acoustic live drums, you can suddenly feel overwhelmed rather quickly. Presented with three Kick's, Two Snares, Hihat, Toms, Overhead and mono-stereo Room mic's, how on earth do you know where to even start. Well fear not whilst there are no rules to mixing, it's whatever fits the song material, there are some definitive tried and tested methods to adhere to. To sum up drum mixing in four words, thump, snap, attack, sustain.
To sum up drum mixing in four words, thump, snap, attack, sustain.
Decisions, decisions, decisions
First thing to lock down in any mix is to decide who is going to rule the low end, the kick drum or the bass guitar. There's so much low end information from both instruments somethings got to give. Sometimes you can tell just by listening to the track. If the Bass guitarist is playing mostly high up the neck then that decision has been made for you, the kick can provide the low end energy for example. In certain genre's there are specific ideas applied. In EDM it's generally the Kick drum that rules the roost as that's what provides the beat and drives the song along getting people up to dance. In thrash metal there's a different approach as the Kick and Bass guitar have to be locked in sync to provide a tight rhythm section for the heavy guitars. In reggae you'll mostly want bass to take the low end.
Everything from setting up a trigger on a gate to duck the bass guitar down when a kick drum hits to what you boost on your kick you duck down on your bass and vice versa with an EQ, can be methods adopted to get the two instruments to stay out of each others way in the mix.
decide who is going to rule the low end
Getting Technical
Your next job, albeit an unexciting one, is to check for phase and latency. Put more than one mic on any sound source and relative to distance between mic and source you can now have phase and time delay issues.
Phase and latency come about from mic's not being equidistant from the centre of the sound source and room reflections. So for example it's common when setting your over head mic's, left & right, to measure from the centre of the Snare drum to each mic and have them the exact same distance away to minimise these problems. It can get quite complex when dealing with phase and latency but fortunately there are simple methods to help correct it.
In Studio One we have the 'Mix Tool' a simple trim fader with a polarity change button. Place that as an insert on a track and check if the sound lost it's bass and body by turning the polarity on/off. For latency, time delay, you can check my previous topic on it here; Dealing with Phase and Latency
Dealing With 'Bleed'
Of course as with all drum parts a gate can be used to cut out bleed but it doesn't necessarily give a natural sound and setting up a gate can be quite tricky and intensive requiring some skill to get it just right. Fortunately there are some very interesting plugins that take all the hard work out this task, most notably Drumleveller and Drumatom.
The decision to cut out the bleed from other parts of the kit will depend on whether your going to compress that kit part (kick, snare, toms all close mic'd) as that will not only raise the level of the kick/snare/toms but all of the bleed from hihat, cymbals and room (distant mic's) as well.
On the other hand the overhead and room mic's is where you do want all of the bleed of the full kit coming through.
Bonzo's late Ludwig 26x14inch Bass Drum
Thump
When you think of thump it conjures up an image of very large speaker cones visibly pushing in and out as the kick drum hit's you in the chest knocking you off your feet, we'll especially if the neighbours are away on holiday. There is nothing like it. But before you reach for one of the many low end frequency booster plugins, Rbass, Lowender etc, don't forget about basic tools of the trade that have just as much power, in this case Eq.
Boosting frequencies, especially adding a nice resonant musical bump around the 35-80hz region with a low shelf using an Eq can instantly add more thump, the secret is to keep a natural sound. You most certainly don't want to introduce sounds that aren't there, that aren't part of the original Kick. But if things still need more help down there then you could add some low end harmonics, that will have more of a natural tone than excessive Eq boosts with things like Waves Rbass, or reFuse Lowender plugins and many others to choose from. Solid and natural is the key.
Next for an Eq move is to bring out the point of the Kick drum or the beater. That's the sound that can push it through the mix usually found between 2-4Khz but can depend on the Eq used. How to define that sound? It's the sound of the beater hitting of the drum head, almost a click, that adds the instant attack of the Kick.
Now generally you can get the sound you want from just using the Kick 'In' mic using the methods mentioned above. Adding in a Kick 'Out' (mic placed further back from the kit) or a resonator (designed for sub bass information) has the unwanted effect of adding more bleed (picking up the sound of the rest of the kit) that using these can be more problematic than being beneficial. So don't be afraid to mute those tracks if you find there just adding stuff that doesn't sound good or adding phase issues (mic's placed various differences from the sound and the time it takes that sound to reach that mic presenting delay and latency). A good way to decide is if the presence of the Kick is very much to the forefront and bleed is more controlled (As in a Drum resonator mic where sometimes a blanket can be placed over it to cut down the bleed from the rest of the drum kit).
Ludwig Supralite Snare Drum 14 x 6.5 in.
Snap
Yes we are talking about the all important Snare. It's possibly the most important part of the kit and matching the sound and tone and blending in just the right amount of reverb can have the most profound effect. But listing settings when there are so many variables is pretty much pointless but in saying that on a Neve 1073 Eq adding a nice smiley face Eq curve by high passing 50hz, adding some 220hz and boosting some 7.2khz and 12k shelf can give you a snappy snare but with a nice round bottom end, a favourite of Mr Andrew Scheps, can't but help. But what might work for his mix may be the total opposite to yours. For instance his drums, you can guarantee, where not recorded in a bedroom or basement but more like a special acoustically treated room with thousands of dollars of outboard gear and mic's. The point being you need to get the best sound out of what you've been presented with and that might be the most ridiculous Eq and Compression settings you ever had to use.
Adding just a pinch 0.5dB of distortion to a snare can make it sound like the drummer is hitting the snare harder and help it cut through more if you find it's still note quite there in the mix.
Classic UAD 1176 compressor
Attack
When it comes to defining attack on drums the first thing comes to mind for most is compression. But is that the best way to go?
Setting the attack time on a compressor is how quickly the compressor kicks in on the initial hit of the drum. Setting a fast attack, the compressor kicks in immediately grabbing the volume level of that initial hit and bringing it down, if it's too loud in relation to where you set your threshold level (At what volume do you want the compressor to start working) or the opposite of bringing it up. Setting a slow attack allows the hit to go through before the compressor kicks in and if you spend enough time getting it just right you can enhance the hit rather than taking it away. After all a compressor is just an automatic volume control. You tell it where you want your volume to be (threshold) and it highers or lowers, in relation to your setting, the sound going through it.
But wait! What about the natural sound of the drum hit? If we over use compression as a direct insert on a snare or kick we are in danger of killing the dynamics (changes in volume) altogether. The strike of the drummer hitting down on a snare can be levelled of and now the compression is more detrimental than beneficial. Sometimes it's better to accentuate that hit with a simple Eq boost that will not change the volume levels but make that instant drum strike more apparent if that's what's needed to cut through your mix. That is where the power of parallel compression comes in. You can leave the natural transients alone, be as aggressive as you want with your setting on the parallel track and subtly blend those in, the best of both world so to speak.
So attack may seem about compression but it's about keeping those lively transients in, let the aggressive natural strike of the drummer breathe and chose to enhance them in non-detrimental ways (harmonics, distortion, saturation, parallel compression) that don't take away, but add to the overall sound.
Sustain
Sustain or maybe better described as decay (how long the sound lasts after the initial strike) can be controlled by a few things in the mix the most obvious is the release time on a compressor, although delay or reverb can be used subtly too. Setting the release time is all important for maximum sustain if that's what your after. Think of it this way when a drummer hits the snare you want the initial hit to have impact and be up front the as that sound dies away you want it to last just until the next drum strike. That's how you decide where to set your release time. The easiest way to do this is to your threshold set, close to final attack set and then set your release time by watching a level meter. Solo the snare track watch the level meter for the initial strike. Then as you set your release time the level meter should show the volume drop almost to zero before the next snare hit. That is a perfect release time. Of course you may want certain effects that your going for so things would differ but that is a great foundation and basis to work from.
That's why for anyone, in the early stages of learning about compression, drums are the best place to start and understand exactly what effect they have on sound.
On a lesser note using delay or reverb can obviously make a sound sustain for longer and works well on both snare and toms. But sometimes just adding in the room mic's and overheads can give you all the sustain you want as they capture reflections of the room your kit is set up in.
Too Trigger Happy
When it comes to drums not having the quality sound we're looking for drum replacement has found an ever popular niche. Now when a Kick, Snare or any other drum part isn't cutting it, it can simply be replaced or have a replacement blended in with the original. Whilst this has become more and more of the norm in music production nowadays it's still maintains a level of controversy for some. If we think back to those classic recordings and performances of our percussion heroes ask yourself how would they have reacted to their sound being replaced. What was it about Bonham, Moon, Paice, Rudd, Collins and so many others that made them so special. It was how they hit those pigskins and how they poured there own personality into the art of keeping the beat and providing the all important rhythm section that made it so special.
On top of that you had the engineer with not only years of experience but ears of experience who had it down to a fine art selecting the right mic and placement for optimal sound quality. Maybe if we spent more time on gaining the skill that these Guy's had, replacement would be null and void.
Slate Trigger
It would seem we've almost gone full circle after complaining that drum machines and drum programs had no life, no soul, dare I say they were to perfect and lacked the human touch? However I feel about drum replacement it's here and it's likely here to stay. The best of these programs are most notably Slate Trigger and XLN Addictive Trigger which make the whole process as simple or as complex as you want it to be.
Starting Level
Coming from the analogue world and now mixing in the box this presented somewhat problematic to me. I was so used to watching VU meters I had to find a way to replicate that in the digital realm. Thankfully a friend pointed me to Klanghelm VU meter It has seriously changed my ability to operate and brought back a familiarity that I love to mixing.
Jacquire King has a method he uses of having the VU meter calibration set at -18dB (OdB on the VU meter). He then aims his solo'ed kick drum to be hitting -3dB on the meter at it's loudest section in the song. Then when he adds in his Bass guitar it should average around 0dB and he build the track around that whilst maintaining these levels.
Benny Faccone likes to have his Kick at -5dB and then with the Bass guitar added in the VU level hit -3dB. Whatever you decide mixing in the box is a different animal than analogue you can't mix the same way, things including levels are completely different. But a simple rule of thumb is to be around the 0db level on the VU meter with Kick and Bass solo'ed and playing back together.
In the analogue world you could have your VU meters slamming on the needle and that would not have proved detrimental to your mix, sometimes it was even preferential when hitting tape and wanting more effect and saturation. But in the digital world, level meters, which confusingly differ so much from Daw to Daw, if they were allowed to peak the same way would prove catastrophic to your mix. Anyone who has heard digital distortion can verify that.
Summary
In conclusion it's plain to see that recording and mixing drums is one of the greatest challenges you'll encounter when you pull up the faders for the first time. But take a deep breath, remain calm, get your kick sounding how you want it, then the snare, toms, all your close mic'd pieces of the drumkit, blend in your Overhead and Room mic's and your done. Of course as you mix the rest of your song you may have to revisit some settings but they'll be minor tweaks just to make something sit a little better or duck down if it's sticking up to much.
Watch the video as I go over some of the highlights of what's been talked about here.
So with the first video on initial gain staging your project, I inadvertently added some confusion, sorry for that. So I just wanted to update everything and add in a little twist (kindly pointed out by others) to make things as simple and straightforward as possible so no-one was left in any doubt as to how to get it done fast.
Still using the Klanghelm VU meter but adding on a little bit extra to aid you in setting it up and getting on with mixing. This can be transported to any Daw as confusion reigns with each Daw having different level meters and different ways of working. Watch the demonstration below:
In the days of yore an old analogue console had a three way switch for panning. That was full right, full left or centre (both left & right). Nowadays when it comes to panning we can have any setting we desire but how does that come across to the average listener? If we go all out in panning things around the stereo image can anyone, apart from an engineer with trained ears, actually hear those subtle differences?
There's still something to be said for the power of old school L-C-R panning in a mix and it's still very much a go to in modern mixes. But in saying that so much has changed as to how we now listen to music, ear buds, mono bluetooth speakers,crappy laptop speakers, that some would argue L-C-R has lost some of it's mojo as it was originally intended for the listener to be sitting at the required distance centralised in front of two speakers approximately 6-8 ft apart to create the optimum stereo image. That way there was no doubt as to something being, guitar solo for instance, panned hard right, as in the classic AC/DC - Back In Black song.
But panning isn't just about deciding what speaker or what side a sound should be heard on in fact it can be used to add spatial transition to enhance your mix. For instance say you have two electric guitars. One panned 60 to the left, the other panned 60 to the right. That still has somewhat of a mono feel to it but when it comes to the chorus you could automate those two guitars to go full 100 right and left and now the width increase sounds to the listener like the whole mix just got bigger and wider adding more interest and excitment when it hits the chorus section of your song.
Panning is more apparent on headphones which give more of an effect, almost a phantom illusion, to the listener as to where a sound is coming from. If your listening on your laptop your most likely not aware of a stereo image at all but that of mono sound. The same for someone not sitting directly centre in front of two speakers 6-8ft apart. If your across the room you won't notice what sound is coming from what side of the speakers just that there's a wall of sound that might as well be mono.
So as you can see panning isn't as straight forward as it used to be, governed now, by how most people listen to music. But we created stereo and at the time it was revolutionary so for now panning still has validity in your mix, just an no more, the question being how many of us actually need stereo anymore? Yet as engineers we over think things and analyse the heck out of things spending hours painstakingly critiquing over our mix decisions for someone to then listen to it back on laptop speakers and not be aware of your panning or any other brilliant ideas you thought you had injected into the track.
Note: Panning full left or right can decrease the volume by 1.5dB or more due to Pan Law that differs from product to product and even Daw to Daw.
Balancing The Mix
This is the part of mixing I love the best. You've loaded in the stems that have been sent to you, sorted them, gain staged them and maybe even done a little panning around. But then it comes to listening for the first time to the song and you start to bring up those faders hearing the different parts, different instruments and you level up how loud things are in context to others. No plugins to worry about, no major make or break a mix decision made, just levelling volumes of individual tracks so everything can be heard at just the right level when everything else is playing back.
This is when you can build a relationship with the song and get a feel for where it wants to go without slapping a thousand plugins on it or using outboard gear. Simply just creating a nice balance and enjoying the track as a listener rather than a analytic sound engineer. Sometimes, depending on the quality of the recording, you can get a truly exceptional balance and panning mix going so well you'll think you have nothing else to do, job done you might say.
Watch the video below as I show some of these ideas and how to possibly implement them.
When you're sent a bunch of stems (audio wavefiles) of someones mix and you've been given the task to turn it into an instant hit, things can seem daunting and a little overwhelming. As you import those stems and scroll through possibly 40 to 100+ audio files how do you know where to even start let alone hand the client back a finished two track mix?
In this short series and rather than showing settings, for Eq's and compressors etc, as so many others have already done, I want to walk you through a mix and offer ideas, some philosophical, some practical and examples as to how you would find your way to the finished song. It's not about the plugins used, it's not about the settings used, as these constantly change from song to song, it's about getting from raw to finished two track songs.
How and where to start
This may seem to be the most monotonous and unexciting of tasks but believe me when I say sorting out stems, grouping, colourising, re-naming and finally gain-staging everything is the most important part of starting any mix project. Celebrity mixers (Chris Lord Alge) have junior engineers doing all the grunt work for him, like roadies setting up for a gig, he has them sorting and setting up the mix the way he likes it ready for him to walk in and do his thang! But for most of us we have to do this task ourselves, more is the pity.
Laying a good foundation at the start will dramatically reduce time spent on a mix and that is a good thing. It leaves more time to be allocated to actually listening to what your trying to achieve rather than being distracted searching for the other Kick Mic tracks or where the Lead vocal is at and seeing levels clipping etc.
So lets dive in as I show an initial setup and preparation for getting ready to mix.
After the recent update release of Studio One version 3.3 alas a few ghosts in the machine, bugs in the code appeared and some had trouble with various issues. Presonus being ever so customer friendly have moved extremely swiftly in releasing a maintenance update version 3.3.1 build 39379 that should apply all fixes.
The update can be downloaded inside of Studio One 3 or by going to your My Presonus account, logging in and downloading the installer from there.
The update specifically addresses the issues below;
Version 3.3.1 Release Notes (September 6, 2016)
The following issues have been fixed:
[Win] Melodyne 2.1.2 causes Studio One to lock up on launch
[Win] Older Win 7 machines are missing a MS dll
[Win] Softube Console 1 fails at launch
[Win] Moving an Arranger section does not move included events
[Win] Plug-in compatibility issue with outdated PACE implementation
[Win| compatibility issues with certain audio interfaces
Preview function for layers not working
Disabled Tracks and instruments lose bus routing
Missing a "Cancel" option in Combine/Load Instrument dialog
Disabled Channels are part of Mix Engine FX signal flow
Disable-state of Plug-ins not saved/restored in presets
Activate monitoring after enabling Track not working
Output routing changed for disabled instruments
Thanks for this Presonus great work, I'm sure you'll all agree.
When it comes to an update release for any software it usually starts with lots of "Oooh's" and "Aaah's" later on there's RUNNING AND SCREAMING! Things can transpose sweetly for some and for others endless mishaps with things suddenly not working as they used to and the inevitable roll back waiting it out for a fix to be issued. So it's good advice to sit it out on the touchlines for a while and watch as others dive in an possibly hit their head on the bottom, so to speak.
Rather than do a full in depth look into the update v3.3 (Which has already been done by others) I thought I would be more personal and say what I found to be Pro's or Con's for my own workflow. So with that, check out the list below and the short video.
Pro's
One click global Fx's On/Off - Now that has been a feature we have all been waiting on. Nothing worse than spending endless hours splashing plugin inserts all over our mix thinking we have created a master piece, taking a break, and then coming back to hear how much we dislike what we thought was brilliant before. Now as we go along we can A/B things, with all our adding of plugin's and see if we making things better or worse or even no different at all.
Combined On/Off for multiple selected Effects and Instruments - This has proved to be a major enhancement in speed to my workflow. Before hand wanting to send from various multiple tracks to say a Bus or Fx channel was ever so awkward, laborious and time consuming. Now as long as you have the tracks you want to work on highlighted adding a send to one automatically adds the same send on the other tracks to, brilliant!
Disable Tracks, Instruments or Plug-ins - For those working on a lower spec computer this is fantastic as CPU hogging VST Instruments, plugin inserts or whole tracks can be temporarily disabled to allow you to carry on working when your CPU wants to give up the ghost and die.
Performance improvements for Mix Engine FX - Console shaper had proved to be a bit of a hog on CPU resources and now it comes with a tweak to lower that drain on power. Now in some plugins doing this can seriously effect it's signal quality but so far Console shaper sounds the same to my ears.
Copy Track now includes automation - Duh well shouldn't it have always? Great to see it's now been implemented.
Con's
Integration with Notion 6 - Not a major complaint, but as I personally don't own or use Notion this was of no consequence to me.
S1 Freezing on startup - Some have experienced S1 freezing on startup, this is mainly due to Melodyne have a few issues. I'm told if you log into your Presonus account there is an update for the integrated Melodyne that fixes this issue, fingers crossed for all those affected.
Soloing Layers Problem - A lot our finding trying to now solo layers in version 3.3 gives no sound, a fix is currently in the making.
New Autofill option for multitrack drum quantization in Slice mode - I can understand why this was added (For Oldschool Pro Tools users) who may be so used to doing things this way but there's far better and more efficient ways of doing the exact same thing in Studio One it kind of makes it redundant to anyone who never worked this way in the first place.
Now that's what update version 3.3 brought to my workflow and overall a most welcome one for me. You on the other hand (depending on your own workflow) may have experienced other issues but as usual updates are system specific, what works for one doesn't work for another. Hopefully the few problems already identified will have a fix released in the very near future.
