The Music Surgeon for Royal College of Surgeons INK

Website launch: 'hitmixing.com' & closure of blogs

2009-10-17T14:25:00.000-07:00

We recently stopped using the domain 'mirrormixing.com' as we're about to launch a brand new commercial recording studio company in the LA area. Due to this and an over-crowded schedule I have decided to stop publishing articles till further notice.

The new website can be found at: http://hitmixing.com/ and will be online soon.

Yours sincerely,

The Music Surgeon

Exciting The Drums Timbre

2009-05-26T10:39:00.000-07:00

Intro

The main focus as a mixing engineer is to get every thing; cleaned up, polished, and give every thing its own space and place so it can be easily heard without fighting for the same space and frequency range, I think I don’t only speak for myself when I tell you this? To me this is the ‘upfront stage’ of the mix. The ‘background stage’ however needs to be exciting with loads of stuff moving around, modulation is key to me when I start adding ‘dirt’ to instruments in terms of reverbs, delays, harmonizing, auto-panning, and so on. Most mixers don’t do these things with drums, drums are foremost static for the one reason that parts of the kit are those that always remain and be expected in our most valuable aural real estate (left – center – right). Ever tried to add a stereo reverb to a kick drum for example? That sounds odd to the human ear and therefore translates best to be kept mono not taking into account that most multiple speaker set-up and its crossovers treat the low-end as mono anyway so creating a so called ‘stereo’ reverb wouldn’t be advisable.

70’s Then The 80’s Came (Entering A New Millennium)

After the 70’s the 80’s came with the introduction of digital which opened a complete new world to the engineers back then. One of the key factors for them was that ability to try out effects and see if it worked or not, easily made and easily reversible. Somehow I feel we lost that touch when we entered the 90’s and completely forgot about experimenting on a large scale in the 2k’s as we’re way to busy battling to get the loudest record (unfortunate). Especially now when real estate prices are increasing and studio rent prices are dropping; playing around and trying new things has become a luxury only available for those at the top which at the very same time are too occupied with work to try this. Why I love my job is to be pioneering and try new things out, make the technology available more musical and above all be creative and artistic.

Phat-Up The Drums Timbre W/ Effects

Add Weight To The Snare Drum: A great technique to make the Snare Drum sound a lot more fatter is to add a gated reverb pitched down a 4th from its original key. Duplicate the original Snare Drum track and pitch shift it down by a 4th from its original key, add a Gated-Reverb effect but without the direct sound of the Snare Drum and blend it in with the original sound. Balance it out towards your own liking.

Add Spark To The Snare Drum: Duplicate the original Snare Drum again but this time hit up an Exciter and dial in around 4,000 (approx. B7) cycles depending on the key of your Snare Drum and set Even or Odd Harmonics, which again is dependable on the key of your Snare Drum e.g.: even harmonics are: c , c, g, c, e, g , bflat, c and sound happy adding brilliance, odd harmonics are: g, e, bflat, d, f#, aflat, b and sound dark. A reminder I have to set here is a general law in using tube gear; many people add tube gear to odd harmonics and claim their product is not working right or it doesn’t do much. Put tube gear on the even harmonics and every one will be happy ;-). Another note to set here is to, and I have pointed this out before but it plays a major role in using exciters, not mess up and confuse harmonics with overtones (!!!). Keep in mind that even harmonics are uneven overtones and uneven harmonics are even overtones. Blend this in with the original Snare Drum to your liking.

Add Weight To The Clap: A great technique to make the Clap Drum sound a lot more fatter is to add a gated reverb pitched down a 4th from its original key. Duplicate the original Snare Drum track and pitch shift it down by a 4th from its original key, add a Gated-Reverb effect but without the direct sound of the Clap Drum and blend it in with the original sound. Balance it out towards your own liking.

Add Snap To The Clap: Duplicate the original Clap again but this time hit up an Exciter and dial in around 4,000 (approx. B7) cycles depending on the key of your Clap and set Even or Odd Harmonics, which again is dependable on the key of your Clap e.g.: even harmonics are: c , c, g, c, e, g , bflat, c and sound happy adding brilliance, odd harmonics are: g, e, bflat, d, f#, aflat, b and sound dark. A reminder I have to set here is a general law in using tube gear; many people add tube gear to odd harmonics and claim their product is not working right or it doesn’t do much. Put tube gear on the even harmonics and every one will be happy ;-). Another note to set here is to, and I have pointed this out before but it plays a major role in using exciters, not mess up and confuse harmonics with overtones (!!!). Keep in mind that even harmonics are uneven overtones and uneven harmonics are even overtones. Blend this in with the original Snare Drum to your liking.

Add Body To The Hi-Hats (Overheads): A great technique to make the Hi-Hats (Overheads) sound with more body is to add a pitch delay pitched up or down (depending on what you want to achieve and what sounds better within the actual mix) by 1 semitone (NO cent!!!) via a dupe of the original Hi-Hats (Overheads).

Add Shine To The Hi-Hats (Overheads): Duplicate the original Hi-Hats (Overheads) again but this time hit up an Exciter and dial in around 7,000 (approx. A8) cycles depending on the key of your HiHats (Overheads) and set Even or Odd Harmonics, which again is dependable on the key of your Snare Drum e.g.: even harmonics are: c , c, g, c, e, g , bflat, c and sound happy adding brilliance, odd harmonics are: g, e, bflat, d, f#, aflat, b and sound dark. A reminder I have to set here is a general law in using tube gear; many people add tube gear to odd harmonics and claim their product is not working right or it doesn’t do much. Put tube gear on the even harmonics and every one will be happy ;-). Another note to set here is to, and I have pointed this out before but it plays a major role in using exciters, not mess up and confuse harmonics with overtones (!!!). Keep in mind that even harmonics are uneven overtones and uneven harmonics are even overtones. Blend this in with the original Snare Drum to your liking.

Reverse Ambience Of Percussion: Percussion always benefit from a little bit of reverb but to make the Percussion part more exciting I always set up an Aux with a reverse reverb algorithm inserted on it. This trick works wonders with a clear eye on the fills, can you hear the effect already in your head? Taking it a step further, add a delay with no feedback (!!!) before the reverse reverb insert and set a reasonable large delay this way your reverse Percussion ambience gets more notice and a more bigger appearance in sound in the mix.

Punch On The Drum Group: Don’t we all love those rock records with the drums cutting through the record, the nice punchy attack response? The “secret” lies within the one magic box manufactured by a company named Empirical Labs; their product “Distressor” is a great dynamic processing tool which offers the user the introduction of added harmonic distortion in combination with a compressor/ limiter. It is capable of adding 0.2% to 20% harmonic distortion to your signal depending on the mode (Dist 2 or 3) and it’s release time. I always use it with its HP (high pass @ 90Hz) detection activated so it leaves the biggest energy catcher alone; the low-end, but focuses more on the attack notes of the signal. Setting the Distressors attack time fairly open with a tight fast release response time enables you to create that nice punch on top of your drums. This trick also works on individual drum tracks, for example the Kick Drum and the Snare Drum.

Sideline Tips

Putting a Harmonizer after the Reverb and/ or Delay insert can create a rich sounding ambience to any instrument’s timbre.
Tempo Delay is, I think, one of the most underrated tools in the mixers tool kit for drums, and provides a way to create complex-sounding rhythms from otherwise simple percussion lines. But it has to be done carefully to not disrespect the producer/ and or artist vision. Delays can be simply calculating using the sum, divide 60,000 by the BPM value, then divide again by the desired sub-beat (4 for a quarter note, 8 for an eighth note, 12 for an eighth-note triplet).
Putting an Exciter on the attack note of the Kick Drum with Even Harmonics would benefit it’s tone timbre as well. The very same trick works on the attack notes of a Bass Guitar.

Waves H(ybrid)-Compressor Plugin

2009-05-18T00:40:00.000-07:00

It has been a while since I posted my last post and I almost forgot to publish the following about the new Waves H-Compressor, H stands for Hybrid. Which I had a little to do with.

Taken from the Waves website:

"H-Comp is a brand new dynamics processor that combines the modeled behavior of transformers, tubes, and transistors, together with the power and precision that only a plug-in can provide. The result is a compressor with capabilities that engineers could only dream about back in the day. H-Comp lends a whole new meaning to the concept of THD: Total Harmonic Distinction."

The idea for this plugin came along about a year ago when I was talking to Gilad Keren the co-

founder of Waves. He referred me to his head developer Amir Vinci. We started talking about the concept he was working on and my concept of creating a more musical compressor resulting in the H-Compressor.

Various issues we resolved in the design were;

Parallel Compression (Dry/ Wet)
Pre-Delay Attack (Punch)
and my concept to implement more musical release times (Release)

Parallel Compression:

Parallel Compression is achieved in the compressor by mixing the dry signal (un-compressed) with the wet (compressed) signal using the Dry/ Wet pot.

Pre-Delay Attack:

Pre-Delay Attack is a technique I some times apply to create additional "punch" to a kick drum and or snare drum. You achieve this by making a dupe of the original track and slightly delay this with 4 to 10ms, using this as the trigger to the original track. This is now implemented in the plugin as the "Punch"-pot.

Free, Host & Whole Note Based Release Times:

This is my very own concept to create a more musical compressor starting of with release times. Its the reverse trick of using a delay calculator as release time calculator as explained many times before on my blog. This is now included for the first time in an actual device, in this case a plugin.

The plugin is definitely some thing to look at for!

http://www.waves.com/Content.aspx?id=9111

Yours truly,

The Surgeon ;-)

Pro Tools Snap Automation V2: ICON

2009-05-06T18:01:00.000-07:00

Hey everyone! Has been a little while but I came a cross some thing which I personally think will be great to post it up on here.

Referring back to my old post about "Snap Automation": http://mirrormixing.blogspot.com/2008/11/snapshot-automation-in-pro-tools.html

After reading please watch the following video as addition:

ICON Snapshot automation from Brent Heber on Vimeo.

Free Online Recording Primer & Free Online Solid State Logic 4000 Series Course

2009-04-20T12:27:00.000-07:00

One of the biggest challenges students have who attend a school or college for audio engineering; they don’t own an SSL large flagship console or have enough time to completely understand the console.

The Recording Institute of Detroit have their Recording Primer curriculum online. Most interesting of course is their training material for the legendary classic console ‘SSL 4000’. The entire course has been written by Robert Dennis.

It is definitely worth a look: http://www.recordinginstitute.com/da154/ARP/

The Search For The Magic Formula

2009-03-25T14:58:00.000-07:00

For quite a bit of years I have a document with me which changed the way I hear music. Many of you know I used to compose music back in the 90's so therefore it's no surprise what I'm about to tell you.

The document I'm talking about is a case study done by D. P. Hyde back in his BA Hons year at the University of Birmingham in the UK. It digs deep into the success factors of production team (Mike) Stock, (Matt) Aitken, and (Pete) Waterman in the mid-late 80's and early 90's. They are considered to be the most successful songwriting, and producing partnerships of all time. Allegedly they sold over 40 million records, including over 100 (!!!) Top 40 hits. However, some key elements to their success can be found in parallel to the famous earlier Motown era.

The typical chord progression used by Stock, Aitken, and Waterman has been used many years before them and is still used; the famous "50's Progression" (e.g. I - IV - V which is actually a ii - V - I). A trick you can use is the "Circle of Fifths". It's also a way to easily and convenient toggle around chords to determine your chord progression.

You probably wonder why am I posting up this blog. The reason given is simple: I think its an essential piece of history and knowledge for every one involved at any stage of the production process in music. To better understand popular music, thus better understanding the music you are working with.

You can download the final study paper here.

Interactive Instrument Frequency Range Chart

2009-03-20T11:19:00.000-07:00

I know I have posted up a instrument frequency range chart before but I came a cross a real nice and detailed interactive one from Independent Recording . Net website. Go have a look:

If you want to invest a bit of money please go and order the chart made by E.J. Quinby named "Musical Pitch Relation Chart".

Signing off,

Yours Truly!

Stand Alone VST/ RTAS/ DX Analyzing

2009-03-20T06:26:00.000-07:00

Over the past few weeks I did receive quite an amount of emails concerning the way I do stand-alone analyzing with plugins. I thought it was time to address this in one of my blogs and here it is. I usually use a combination of Mac/ PC, meaning Pro Tools HD running on the Mac and the stand alone VST shell on the PC running Linux. In the example used: Windows. The main reason is to keep this simple for now.

I'm using a program called "Console" made by a company called "ART Teknika" and it's retail price is approximately $54 bucks. If I do stand alone analyzing on a MAC system I use a program called "Ardour" via the use of "JACKaudio" to inter connect all the I/O's which allows me more accurate things, but I think "Console" will work for most of you on a 'need to know basis' for the time being.

Here's a screen shot of "Ardour"

Coming back to "Console". "Console" is very easy to understand and work with you simply set-up your audio card and select the VST/ RTAS/ or DX directory or directories where the plugins are located and "Console" loads them into it's shell ready to be used. The next step is to set up your routing and signal flow. Drag down your sound card's I/O and drag down the desired plugin in between and hold shift and click from left to right until all the modules are selected, once selected right click and select "Audio Link". You can't get it more easier than this! To avoid a feedback loop make sure you click the switch on the audio output. Make sure you make up for latency as well.

To find out more about "Console" please visit their website here.

For any questions please email me at royalcollegeofsurgeons @ gmail . com

Signing off,

Yours Truly!

Waves Center: M/S Compression made easy

2009-03-20T05:19:00.000-07:00

To get back to my earlier post "Advanced Compression'; plugin developer Waves created a beautiful plugin which incorporates the concept of M/S Compression in an easy GUI and it's named "Center". Go check it out!

Yours sincerely,

The Music Surgeon

Advanced Compression (Painting The Picture Of Music Book Excerpt)

2009-03-15T09:48:00.000-07:00

Advanced Compression

Intro
It took me a minute to get this blog together as it actual represents a real chapter in my yet to be published book “Painting The Picture Of Music”. Most of you, the readers, will find some information which has already been published earlier. For example M/S Compression and Dynamic-EQ. However this blog will go a bit more deeper into the magical, but above all, beautiful world of side-chaining. I guarantee you that you can’t live without it by the time you have finished reading this blog.

Over Compression
You all have read about this earlier in one of my blog post. But today I’m talking about the sense of setting a lower threshold than you need to get the job done. I though to start of with this before I even dig into advanced compression/ side-chaining.

As you know over compression will always make the sound worse, with the sole exception of percussive sounds (!!!) where it some times be a useful effect. Imagine for example that you have a sound source that plays occasionally but with silences in between. This is where usually over compression is best audible and most likely to happen. When you set the threshold, most likely you know where you’re aiming at of how much gain reduction you want to apply; hear and both say on the meter. This is controlled as we know by the ratio and threshold controls. Lets say we’ve set it in a way that we get 10dB of gain reduction. When the sound source passes through the compressor ask yourself this: “Does it ever go all the way back down to zero?”. If this is not the case and it only goes down to 3dB then you haven’t applied the set amount of gain reduction, in this case 10dB. What you did achieve is 7dB of compression reduction. The other 3dB could have been achieved by simply lowering the fader of your console. The problem starts when the sound source starts to play again. What happens next is that your compressor has to go from unity gain (0) all the way down to the full 10dB. The 3dB we just talked about, that the compressor also has to cover, will result in audible distortion from the compressor and distorts the initial transient. This brings us to rule number one of compression: “At some point in the track, while the audio source is still playing, the compressors’ gain reduction meter must indicate zero, otherwise the minimum reading obtained shows wasted gain reduction and thus over compression leading to the distortion of transients that follow the silences.

High Level Compression
This is most likely the best known usage for a compressor; to increase the apparent loudness of an audio signal. It is called high level compression as it works by reducing the high signal levels (peaks), bringing them down and closer to the low level passages. After this is achieved you apply make up gain to equal it’s input.

The danger however in this technique is that it is very audible and there should be great care in setting up the compressor and compromise between getting enough compression going and not to spoil the overall sound.

Low Level Compression
Ray Dolby is a great example to start the subject of low level compression off with. Roy Dolby told us back in the early A-Type noise reduction system that he left the high levels completely alone and modified the gain only of the signals that fall below -40dB. Every ordinary high level compressor is capable of doing low level compression. A low level compressor will bring up (expand) the audio signal that falls below the set threshold and leaves the high level content completely alone.

Parallel Compression
Another way of doing this is making dupes of the specific audio track. This technique is most commonly known as parallel compression. What you do is mix the uncompressed signal with the compressed signal. At the levels below the compressor’s threshold the two signals will combine to produce a 6dB increase in level. Above the threshold the compressed signal will be progressively reduced and add hardly any additional level to the mix. The outcome is a form of compression where you can get more dynamic range reduction as a result with fewer audible side effects, thus it will sound more human.

Please note that by applying this technique in the digital world you have to be careful of delay issues involved in digital processing, and compensate for the delay else you’ll end up with a mess of comb filtering.

Compression vs. Clipping
We have just talked about over compression and its distortion over it’s long term processes of working at the very least over periods of tens of milliseconds. If however you try to achieve very fast compression by using very short attack and release times, it will result in audible distortion of the low frequencies. What happens is that the compressor changes the actual shape of the waveform and of course you can push the compressor to its very limits up to the point where it has given all it has got to give.

Clipping however does work on a very short time scale. Think about transistorized circuitry which operations are based on the microseconds to any level that is too great for the power supply to cope with and cuts it short, resulting in harsh harmonics, which in addition results in apparent loudness and is called “soft clipping”. What actually happens is that the soft clipping of valve and valve emulated designs rounds rather than clips when dealing with the peaks. When used alone it results in the problem that it will only be affective on high level signals however clip-worthy peaks occur but in high and low level signals. To make full use of soft clipping I highly advice to use a compressor pre and the soft clipper post (use them in series). The reason for this is that compressor evens out general level of the signal, because the compressor works over a comparatively long period time, the peaks re not clipped but simply reduced to a more uniform level. Therefore the soft clipper has more material to work on. To go a step further add this signal to the unprocessed version and apply equalization to the processed version and select the frequency range that will be affected to add just the right bit of distortion without going into the extreme (mid-range).

M(id)/S(ide) Compression
This is one of the subjects I covered earlier in one of my blog posts. For those who missed it here is it again:

We are all familiar with what M/S does and where it stands for (Mid/ Side). M/S compression will give you another angle to how you will tread your stereo buss (mix buss). The M channel is the whole sum of the song, and the S channel represents the difference between left and right.

So M/S compression lets you compress and control the center, and sides of your stereo mix separately. This, all the sudden, allows you to bring up your vocals without affecting your instruments, bringing them back. Taming kick and snare on your overheads, or even emphasizing wide early reflections in the stereo field.

Example: A lot of times you've finished a mix, however sometimes the vocal gets slightly buried when the backing track (instruments) get loud. If you go on and try to compress the overall mix, or by the use of narrow band compression of the vocal's frequency range, you will notice that the compression will ruin the great sounding backing track (instruments). M/S compression will come in handy during this stage. M/S compression can help to isolate the center image (M channel). How? By only compressing the M channel, bring up the center when the signals get loud, or another option is to compress the M channel and expand the S channel. This way you help to control the vocalist and open up the specific band. By compressing the S channel only, anything panned center is unaffected and the compression only affects signals panned left and/ or right that are out of phase. Loud signals in these modes will cause a momentary reduction in level of the S channel and therefore resulting in a narrowing of image width. Another option is multi band M/S compression but I'm not going to touch that option today as it offers more possibilities. If you do work in analogue I strongly recommend to use a stereo compressor (which side-chains are linked, the reason for this is that you don't want any phase-shifts happening or an imaging change. In L/R compression it's not guaranteed absolutely zero, as analogue compressors will not 100% handle both channels absolutely equally and therefore some degree of image shift and phase shift might occur. However in M/S compression, any disparity between the channels will not result in any degree of image shift, but in a variation of the width of the stereo image which is less obtrusive than the degree of phase and image shift occurring in L/R compression. I personally would say it works way better than L/R compression and you will find that out for yourself once you've played around with it a few times.

To achieve M/S compression simply pass the M signal through one channel of a compressor and the S signal through another. Plugin wise you have to this in two stages.

Advanced Side-Chaining

How To
Let me first start of for those who are fairly new to this technique, how it is achieved. To perform dynamic-equalization (frequency conscious) you’ll need to patch an equalizer (doesn’t matter what quality) into the side-chain of a compressor (parallel a signal so it enters the normal input of the compressor, and at the same is connected to the side chain input via the equalizer). The audio signal you want to process gets patched through the I/O. What you do next is make the compressor more sensitive to the frequencies you want it to dip. On the equalizer you select the frequency or frequencies or bandwidth you’d like the compressor to dip, instead of cutting you give them a boost. The reason for boosting is simply: when the compressor hears an exaggerated response in frequency, frequencies or the band, courtesy of the EQ boost, it will exceed the compressors threshold and make the compressor reduce the level of the audio signal’s specific frequency, frequencies or band. You’re compressor has just become an incredibly flexible and creative EQ. Makes sense right?

The release time of the compressor has to be set right in order to let the compressor attenuate the boosted frequencies, courtesy of the EQ, by setting up the release time so that it releases and returns the track back to unity gain (0) immediately after the frequency has ceased. The attack time has to be set around 50 musec (.05 ms) and an release time of 50 to 60 ms will get the compressor in and out fast enough to attenuate the frequency portion, but leaving the rest of the signal untouched.

Soft Knee & Hard Knee
Lets go beyond side chaining as known to human kind and introduce something very interesting: ‘Distortion Triggering’.

Every one is familiar with compressors soft knee and hard knee settings which in basic words go from immediately (hard knee) from uncompressed to compressed at the exact set threshold rather than the gentle bending of the soft knee types. The knee type is an essential factor in its sound, however very few compressors allow you to modify the knee curve in any way and even if they do it’s a pre-fixed setting for either hard or soft not even a mixture of both which in cases such as parallel compression are very desirable.

So is this it you may wonder? No, there is a trick that enables you to change and control the knee curve of ANY compressor which has a dedicated side chain input which most compressors do have. Instead of hooking up your equalizer hook up a distortion box (e.g. guitar stomp boxes). You won’t hear any of the distortion sound as it’s not in the actual sound path but it functions and is hooked up as a trigger. What happens in the circuitry is that the distortion box sends soft and/ -or hard clipping into the compressor as a control value. This clipping will bend the shape of the knee curve of the compressor itself depending on it’s operation mode and type: peak/ RMS detection. It will result in a way more musical and different sounding compressor that you’ve ever heard. Cool trick eh?

Pre-Delay Compression
Another option for the side chain is to insert an advanced version of the signal to control the level of the signal itself. Hardware compressor can never anticipate or prepare for what is going to happen, they base their reaction on what information is coming in. With this trick it allows you to prepare the compressor for what is coming.

What you do is simply make a dupe of the track you want to compress and shift it in time with respect to the other track. Connect this advanced dupe to the compressors side chain input (delay of around 50 ms should do the trick) and the delayed version to the normal input of the compressor. What happens next is that the compressor reacts more smoother and musical to the transients of the sound being processed, more realistic. You can also do this the other way round which will be very useful for percussive instruments which depend of their transients and therefore require a slow attack time so that the initial transient come through unaltered before its body gets compressed by the compressor.

Pre-Attack Compression
This trick works exactly the same as the Pre-Delay trick above but now we focus on the initial transient attack information and delay the signal only by around 10 ms resulting in additional punch to percussive instruments such as kick drum, snare, clap, and so on.

Vocal Sibilance Example
For example lets create a so called “De-Esser”. We patch in an equalizer to the compressors side-chain. The quality of the equalizer doesn’t matter as it won’t be in the audible stereo path. Usually I start by boosting the area above 5,000 cycles (5kHz). You can do some broad strokes as the energy above the sibilance band is usually not enough to unintentionally trigger the compressor. This is also another way so I don’t have to fuss with the high-frequency cutoff, and it insures me that all the sibilance’s will be attenuated. On top of that I’ll low cut everything below 5,000 cycles to increase the compressors sensibility above 5,000 cycles and decrease them below 5,000 cycles.

To get to the compressors ratio’s I usually start of with u:1 (unity to one) so I can really hear the effect and then back off the compressor to where it sounds right.

High Pass Filtered Side-Chain
High pass filtered side-chain is most commonly used to reduce the influence of low frequency instruments on the gain reduction circuit. For example kick drums, percussion, and so on. To achieve this you set the high pass filter of the equalizer up to 500 cycles. You’ll notice that your drum kit will sound more open.

Subject Of Thought: Compressor Ratios & Release Times
What ratio should I use? That’s a common question. However I’d like to forget about thinking in ratio’s, why? Cause it’s music, it’s something spontaneous. It’s creative. For example when we talk about ratio we talk about the ratio of which the compressor compresses the signal beyond it’s knee curve, e.g.: a compression ratio of 2:1 and a 10 dB increase of input level, will result in a 5 dB increase in level at the compressors output (simple math: 10 / 2 = 5). But is this the right approach? Assume that the threshold is subject to the knee curve which leads to logarithmic compression. HOWEVER beyond this point the compression lessens and the curve reverts back to a straight line, leading to no compression. Transients usually cause problems with compressors, resulting in a compressor going completely out of line and uncontrollable for a short period of time. The answer: Why not letting the transient pass through so we can focus on controlling the steady-state of our audio signal? We (peak) limit after compression anyways to control the initial transient. Add on to this musical release times in whole and dotted note values? Fortunate enough I’m already accommodating these issues with a major plugin developer.

Embrace your creativity!

The Music Surgeon

Penny & Giles: All You Need To Know About The World Famous Fader Family

2009-03-05T17:42:00.001-08:00

Today I was talking to Penny & Giles and their official dealer in North America; Manquen. They pointed out a very interesting section on their website which describes how P&G Faders are made, their testing procedures, the parts and their numbers, and the Flying Faders automated fader system.

I thought it was pretty cool. Here's the link: http://www.manquen.net/png/

Yours Sincerely,

The Music Surgeon

The Important Role Of The Attack/ Decay Transients In Music Instrument Recognition

2009-03-04T14:03:00.000-08:00

Intro
It has long been known that the onset of a note, the attack, plays an important role in our perception of timbre. The attack transient components in music instruments have been known to contain a vast amount of information about the music instrument.

There are hundreds of different types of music instrument on the surface of planet earth today. Each of those music instruments has its own unique characteristic that distinguishes it from another. Looking at the music instruments signal from a temporal perspective we can divide the signal into four main elements; attack, decay, sustain, and retard (release). The main area of focus for instrument recognition has always been the so called ‘steady-state’ (sustain). The reason for this is that’s easy to analyze.

It needs to be addressed that doing a study about the attack transient is very difficult due to its short period of time and its rapid changes. Therefore it comes as no surprise that attack transients are not well understood nor well represented within any of the available analysis-resynthesis models.

Frequency Analyzers
The first step in understanding your sound as a graphical representation is having the right frequency analyzer. As we all know from my earlier post about the frequency analyzer is that it truncates the data (audio) coming in. Not only is this a problem when we try to analyze attack transients, the other important part is the response time of the frequency analyzer which is at a fixed rate also named ‘hop-distance’. It is the analyzers primary task to detect and locate the so called ‘partials‘ or in better understanding language: the sinusoidal elements that compose the harmonic structure of a pitched sound. Therefore to gain the highest possible frequency resolution it is better to have a long analysis window, but for time resolution this is the complete opposite.

Because of the rapid changes of the attack transients –the note’s onset, and the analyzer’s relative slow response time (frames), it will create distortion or in the worst case a drop-out on synthesis as its algorithm can’t handle the amount of incoming energy from the percussive attack transients and therefore create artifacts upon synthesis such as phase discontinuities; which happens in the high frequency spectrum.

Transient Designer As Re-Tuning Device
The resynthesis of the attack transient results in a better perception of pitch. Unfortunate enough NO ONE has adapted a study into this, but you can use this little bit of knowledge in advantage during mixdown. Think about it.
Having that said it is thus possible to ‘re-tune’ audio programs with a dedicated transient designer (Imagine applying (pre-)EQ to this).

Outro
However during the mixdown phase it’s quite uncommon to see a mixing engineer grab for a transient designer to resynthesis the audio program. The reason for this can be found in our tools, starting off with our frequency analyzers. Another important part is the role of synthesizer programmers in this issue. As many engineers still see transient designing or envelope generating as a task for the ‘MIDI guys’.

Signing off,

Yours truly

Freeze Audio Tracks in Pro Tools

2009-02-26T18:37:00.000-08:00

Intro
Quite a lot of times I ran into the discussion when two assistant engineers argue about the fact that there’s no (physical button) freeze function within Pro Tools, meanwhile there is for native sequencers such as Cubase, Logic, and so on. Let me first off correct every one with the common mistake of comparing Pro Tools which is a high definition audio recorder, with MIDI sequencers such as Cubase, Logic and so on; and the answer has already been given ;-). I won’t dig deeper into the entire structure of the differences software programming wise.

Now we have that clear lets dig a little into the history of the process of recording audio and its audio recording devices.
Back in the day when we were in the pre- pro tools days everything got recorded to tape. Which, again, came in different types, sorts, and brands. Best known examples are the ½”, ¾”, 1” and 2” tapes and 2, 4, 8, 12, 16, 24, 32 and 48 track machines.

Whenever mix engineers are satisfied with their final mix, they will go ahead and proceed to print the mix. Usually in different versions like ‘Main Mix’, ‘Instrumental’, ‘T.V. Version’, and ‘a Capella’. Not such a big problem if you don’t print any stems or single instruments, but what did the recording engineers do? They needed to record every single instrument to a different track. As you can imagine running a tape machine with only 4 tracks or 8 would be a big challenge as recording engineer. Despite the fact that you could sync several machines together in a chain, it would still not really give you a solution. Legendary Beatles producer George Martin used a bouncing technique which we can identify now as the early beginnings of the famous ‘freeze’ function in our MIDI sequencers.

Bouncing
Let me give you more information about this ‘bounce’ technique which will clarify its parallel in todays working with Pro Tools as audio sequencer (recorder). At any given point in the recording process, any number of existing tracks can be "bounced" into one or two available tracks and then erasing the original tracks, making more room for more tracks to be reused for fresh recording, or in computer recording to free up more ‘voices’ and processing power (CPU or TDM).

You can see Pro Tools HD as a multi track recorder and its sample rate as the type of machine.

48kHz – 192 Track/ Voice Count
96kHz – 96 Track/ Voice Count
192kHz – 36 Track/ Voice Count

*Note: Set DAE buffer size to 4 for 192 tracks@48kHz sample rate.

Running at 48kHz, you’ll have a 192 Multi Track Recorder, running at 96kHz you’ll have a 96 Multi Track Recorder, and on 192kHz you’ll have a 24 Track machine. But keep in mind that Aux Input Tracks do not eat up any voice(s)!!!

Personally I run all my sessions with a bit rate of 24 @ 96kHz.

Tape Bounce vs. Track Bounce in Pro Tools
To get back to the point of my blog post; freezing tracks in Pro Tools.
Instead of having a physical button for freezing tracks, you record these in a similar way as the analogue tape days to a new audio track and making the ‘bounced’ tracks ‘inactive’ as if you erased them on the tape machine, freeing up additional voices and processing power (CPU or TDM). The advantage of doing this in Pro Tools nowadays than on tape, is that it is not destructive as it was back in the day. By erasing your bounced tracks you lost them forever. By making those bounced tracks inactive in Pro Tools you can always go back in time and make adjustments to those individual tracks. Instant recall at a glance of your finger tips!!!

On top of that puts bouncing tracks less load on your system than if you’d ‘Bounce To Disk’ for instance. Also the Pro Tools mixer works in 48 bit, bouncing this data to disk would occur in the problem that Pro Tools needs to truncate to 24 bit or even worse (when working) in 16 bit (!!!). Therefore your mix will sound slightly different and a bit rough around the edges when using the ‘Bounce To Disk’ function within Pro Tools. You won’t have this problem when ‘freezing’ tracks.

Another advantage over ‘Bounce To Disk’ is that you can record all your stems at once instead of doing this stem by stem.

Setting Up Track Freezing
Instead of routing your stereo mixdown to the outputs 1 – 2 you route them to separate busses (Any available). For example; the aux input track picks up busses 1 – 2, and its outputs is routed to busses 3 – 4. I Solo Safe these Aux Input tracks by holding command+click (MAC) or windows+click (PC). A new audio track is created afterwards and put in Record Mode and picks up busses 3 – 4 and records the audio signal coming from its input. I do activate the ‘I’ (Input Only) button next to the ‘R’ (Record) button so I can hear the entire mix. That’s it!

*Note for Pro Tools LE users: You need to put the ‘bounce’ track in Record Ready, and Input Only monitoring should be selected.

Exporting The Stems
Now we know that ‘Bounce To Disk’ truncates all mixer data, how can we export our stems or even our final mixdown?

Select in Pro Tools in the ‘Edit Window’ the audio regions you wish to export (I hope you bounced entire tracks and not sections, if not please consolidate your audio tracks: Option+Shift+3) and click on the right hand menu in Pro Tools named ‘Regions’ (if this doesn’t appear, click below in your window on the right hand side in Pro Tools on the << face="arial" style="font-weight: bold;">

Outro
Nine out of ten times I can adjust the audio into what the client wants to have changed simply by sub –or additive equalization. For example quite a lot of times clients want to change the overall Snare Drum in the mix cause it’s either overpowering the mix or exactly the opposite. By simply pulling down a little around 4,000 cycles on the EQ on my Drums Stem I achieve a less overpowering Snare Drum, or in the other case boosting a little around 4,000 cycles will bring the overall Snare Drum sound up a bit more in the mix.

This is more time efficient, and way easier than going back in the mix completely via recall sheets and so on. On top of that will my session be extremely organized with all the instrument tracks, dupes and mults inactive, with only the stems active.

You can see that I absolutely never use ‘Bounce To Disk’ anymore.

Signing off,

The Music Surgeon mixer par excellence.

Pseudo-Stereo

2009-02-14T11:50:00.000-08:00

Today I want to talk about the subject of 'stereo' or better addressed the subject of 'pseudo-stereo'. It is a safe bet for me to say that literally all the mixes I get in have tons of pseudo-stereo tracks in the sessions. As many of you know I started out as a composer/ producer of dance music, and as ignorant and young as I was, I also made this common mistake of always tracking the synths and keyboards in 'stereo', as of course the outputs of every synth and keyboard come as a stereo pair. But let me ask you this: "Is what I'm recording really stereo?". Please keep this in your head and think about this for a minute before you continue reading my blog.

We have been accepting 'pseudo-stereo' for decades and have blindly been recording 'pseudo-stereo' sound sources, and we continue doing this with every stereo instrument feed and all our effects returns up until today. We automatically think that it will work out at the end, but the truth of the matter is that we will end up with one big collection and compilation of mono (!!!).

To make sure whatever you track is stereo please be advised to listen closely to the sound source and determine yourself if it is true stereo. Then ask yourself the following question: "Would it work better to separate them into two mono parts?" (And NO stereo is not better than mono (!!!)).

Nearly every time I'm mixing a track I will separate my stereo feed into two mono parts and play around with it's timbre and panning. The reason for this has just been explained above but other than that It gives me more flexibility and creative freedom to play around with the instrument's position in the mix, add depth and movement. This way I won't fill my sacred and most value real estate aural territory (which are; hard left, center (where we already expect our, kick drum, snare drum, bass, and lead vocal to be, and hard right) with every 'pseudo-stereo' instrument feed but create space for the essential instruments and feeds.

By playing around and changing the instruments timbre separately from each others part, you will create a stereo signal at the end and it will be more interestingly sounding then you would expect as a listener.

You will see the more experienced you get in this trick the more interesting and better your mixes will sound.

Signing off,

Yours truly

RE: MAGIC EQ FREQUENCIES: HOW TO LEARN EQ

2009-01-14T19:12:00.000-08:00

Last night I just realized that I have forgotten to give away my biggest secret. That of how I got to know what to cut or boost and what makes an specific instruments sound that way -known as the instruments sweet spot(s). In basic words how to use an EQ.

My best advice for every one that's (starting) learning EQ is to buy a 31-band, 2-channel (stereo) graphic equalizer (it doesn't have to be an expensive one [!!!] It's about the idea and learning curve, but to give you my honest advice I'd invest in an dbx 2231, 1231, or 231) and place it between your hi-fi's or computer's soundcard's output and your speakers.
Now when you play music try and play around with the bands, adding, subtracting, and so on. Listen how the song sounds and alter it in a way you want it to sound, the way it's most pleasant to you. Try to find out the sweet spots of the instruments within the song so it stands out above the rest of the track and vice versa.

NOTE: It's highly recommended to A/B between the EQ'd signal and the non-EQ'd signal to know what you are actually doing to the audio for the better or the worse(!!!).

This is the way I learned and self-taught how to use EQ and I still use it from time to time, very handy! So next time you go into a mix you know what frequencies to aim for ;-) It sure helped me a whole lot!

Just to let you know that I don't do everything by the traditional 'ear'; in the studio I have Waves PAZ running as stand-alone plugin using a RTAS Shell on a separate TV screen. It's inputs listens to the main outs from Pro Tools using the free JackAudio application and a VU kinda response time (closest to the human hearing). This way the analyzer is always shown.
It's a save bet for me to say that the analyzer is probably the most handy tool during mixdown for me in a way that I can see exactly what the instrument is doing in the frequency spectrum. And it sure saved my ass a lot of times. However I can't stress how important it is to STILL rely on your hearing (!!!) the old fashioned way. The ways an analyzer works is not a complete save bet (blindly) as explained in one of my earlier post.

Learning how to EQ will prepare you for the biggest part to later on understand how to transient design your instruments in a corrective way. Also compression plays an important role in this.

Have fun playing around with your EQ and let me know if this worked for you as it did for me.

Scalping off with surgical incision and precision,

Your Music Surgeon

CAPTURE CLIPS AUTOMATION SSL J9K SERIES CONSOLES

2009-01-13T19:30:00.000-08:00

Intro
Capture Clips is an innovative feature on SSL J9K series consoles, which allows the system to capture audio 'clip' information from the console's channels, via the opening and closing of gates in the channels' dynamics sections. Once captured, an on-screen representation of the audio can be used as an aid to the editing of automation data. Capture Clips makes your life easier to clean up noise on audio program's, isolate those tomes that happen once or twice in the track, or even FX tracks including one time hitters, among many more advantages. I personally prefer it over Pro Tools 'Strip Silence' function.

Set-up
First you need to set-up the gates on the channels which have the necessary audio coming through them. The system assumes that audio is present when no LEDs are lit on the gate meter.

NOTE: If this is your first experience with Capture Clips, we suggest you read through this section before proceeding, so that you are aware of the potential benefits of this innovative function.

The Steps

1.) Select: Mix-Desk > Events, and click or stab on the 'Clips' box at the top left of the Events List. This results in two new boxes to the right named: Capture and Clear.

2.) To select the channels you want to capture the clips on, either click or stab on Capture. Now the channel selector pop-up will appear. To select a sequential range of channels, simply hold down the fader status button of the first channel and press the button on the last channel.

3.) Hit play in Pro Tools.

4.) When you hit stop in Pro Tools you will be given the option to Save or Discard the captured clips. If you save the clips, select the Mix-Desk > Overview display to view the captured audio for each channel represented by a series of blue blocks.

5.) If you subsequently wish to discard the clips on one or more channels, select Clips followed by Clear on the Events List. This again will call up the channel selector pop-up. Choose the channels on which you want to clear the Captured Clips information. When done hit OK.

NOTE: Clips will be captured from the first time you hit play in Pro Tools after the Capture Clips function is armed. Make sure that Pro Tools is located to the correct start point before you select channels as described before. To avoid capturing excessive material, you should start and end the run in silence. The simplest way to do this is to select a SOLO button on an unused (empty) channel. However, if you don't start and end with the gates closed, the system may assume that there is a clip running from 00:00:00:00 up to the start of the track, and another clip running from the end of the track to 23:59:59:24 or 29 (!!!).

Outro
The K Series computer's mix system provides a set of automation modes which are specifically designed for use with Captured Clips. If you are lucky enough, as I'm often, to work on a K series 9000 console cleaning audio is a bit easier.

Signing off,

Yours truly!

MAGIC EQ FREQUENCIES

2009-01-12T18:32:00.000-08:00

Before I'm even going to reveal my magic EQ frequencies, I want to take this opportunity to give every one a little advice; I'd rather advice every one to cut before boost. EQ in general works better "subtractive" than "additive". (low) Shelving before compression will help your compressor to act smoother, and cutting before boosting will help you to create a better overview of your mix. Keep this in mind or it will haunt you for the rest of your entire career and probably also your life.

I strongly advice you to sweep around the following frequencies and play around with the amount of subtractive and additive dB's. Keep in mind that you can apply more dB/ octave subtractive than additive (!!!). These issues have been addressed in an earlier post which you can find here.

For some of the subtractive work I make use of side-chaining in the form of dynamic-eq. The reason for this is that it sounds more transparent rather than static.

THE MAGIC CHART

40 CYCLES
Reduce to increase overtones and recognition and tighter sound of kick drum. Shelf equalization.

64 CYCLES
Increase to add more fullness to kick drum (fundamental, 1st harmonic). Peak equalization with a Q of about 1.3.

72 CYCLES
Increase to add fullness to bass. Peak equalization with a Q of about 1.3.

100 CYCLES
Increase for fullness to floor toms. Peak equalization with a Q of about 1.3.
Increase for warmer sound of piano and horns. Peak equalization with a Q of about 1.0 for piano, and 1.3 for horns.
Increase for warmer sound of strings. Peak equalization with a Q of about 1.0.
Reduce to decrease boominess of vocals. Peak equalization with a Q of about 1.0.

120 CYCLES
Reduce to increase clarity on all instruments except, kick, bass, toms and other low-end related instruments. Shelf equalization.

128 CYCLES
Increase for harder sound, clarity and punch to kick drum (2nd harmonic, 1st overtone). Peak equalization with a Q of about 1.3.

200 CYCLES
Increase to add fullness to snare, guitars, and vocals. Peak equalization with a Q of about 1.3 for snare and guitars, and 1.0 for vocals.
Reduce to decrease gong sound of cymbals. Peak equalization with a Q of about 1.0.

240 CYCLES
Reduce to decrease muddiness of vocals. Peak equalization with a Q of 1.0.
Reduce to decrease sustaining sound of bass. Peak equalization with a Q of 2.0.

300 CYCLES
Reduce to decrease muddiness of mid-range instruments. Peak equalization with a Q of about 1.0.
Reduce on kick drum for clarity of mid-range instruments such as vocals and pads. Peak equalization with a Q of about 1.0.

500 CYCLES
Reduce to decrease cardboard sound. Peak equalization with a Q of about 1.0.
Reduce to decrease ambience on cymbals. Peak equalization with a Q of about 1.0.

600 CYCLES
Reduce on kick drum for clarity of lead vocals body. Peak equalization of 1.0.
Increase to add guts/ body to lead vocal. Peak equalization with a Q of about 1.3.
Trick: reduce on all backing track instruments to achieve clarity, and put lead vocal solid on top of the backing track (instruments).

740 CYCLES
Reduce on snare to increase clarity of over-heads, hi-hats and cymbals. Peak equalization with a Q of about 1.0.

900 CYCLES
Increase for clarity and punch of bass. Peak equalization with a Q of about 1.3.
Reduce to remove cheap sound of guitars. Peak equalization with a Q of about 1.3.

1,000 CYCLES
Increase for body of keyboards/ synthesizers. Peak equalization with a Q of about 1.3.

1,500 CYCLES
Increase for clarity and pluck of bass. Peak equalization with a Q of about 1.3.

2,500 CYCLES
Increase for attack of snare. Peak equalization with a Q of about 1.3.
Increase for more attack of electric and acoustic guitar. Peak equalization with a Q of about 1.3.

3,000 CYCLES
Increase for more clarity and harshness of lead vocals. Peak equalization with a Q of about 1.0.
Increase for more attack on low piano parts. Peak equalization with a Q of about 1.0.
Reduce to increase breathy, soft -sound on background vocals. Peak equalization with a Q of about 1.0.
Reduce to increase overtones of bass, and clarity of other instruments. Shelf equalization.
Trick: reduce on all backing track instruments to achieve clarity, and put lead vocal solid on top of the backing track (instruments).

4,000 CYCLES
Increase for attack of kick drum. Peak equalization with a Q of about 1.3.
Reduce to decrease harshness of electric guitars (rock guitars). Peak equalization with a Q of about 1.0.

5,000 CYCLES
Increase for presence of vocals. Peak equalization with a Q of about 1.0.
Increase for attack of toms. Peak equalization with a Q of about 1.3.
Increase for attack of piano and (acoustic) guitars. Peak equalization with a Q of about 1.3.
Reduce to make instruments appear more distant. Peak equalization with a Q of about 1.0.
Reduce to remove cheap digital sound of reverbs and other digital effects. Shelf equalization.

7,000 CYCLES
Increase to add sharpness/ bite on synthesizers. Peak equalization with a Q of about 1.3.
Increase to add sharpness to piano. Peak equalization with a Q of about 1.0.
Increase to add sharpness to guitars. Peak equalization with a Q of about 1.3.
Increase for dull singer. Peak equalization with a Q of about 1.3.
Reduce for less "S" sound on vocals. Peak equalization with a Q of about 2.0.

10,000 CYCLES
Increase for air on vocals. Peak equalization with a Q of about 1.0.
Increase on overall stereo mix (2TR) to add air and brighten up the final mix. Shelf equalization.

14,000 CYCLES
Increase to make sampled synthesizers sound more analogue. Peak equalization with a Q of about 1.0
Increase for air on cymbals, strings, and flutes. Peak equalization with a Q of about 1.3.

16,000 CYCLES
Increase for air on vocals. Peak equalization with a Q of about 1.0.

Note: Please keep in mind that the following are approximate values and are song dependent and root key related (!!!).

I hope this chart will help you to create better and cleaner sounding records. And as for any thing you can always reach me on royalcollegeofsurgeonsinc @ gmail . com.

Signing off from Studio A at Glenwood Place Studios in Burbank, CA.

The Music Surgeon

Looking for extraordinary Sing and Dance talent (Universal Music Korea)!

2008-12-28T13:24:00.000-08:00

Universal Music Korea (Asia) in partnership with MirrorMixing Entertainment Group (MMEG) International (my company) is currently on the look out for the next sing and dance sensation in Urban/ Pop music. The requirements are: You need to be between the age of 13 and 16; of Asian decent; an incredible singer and dancer, and you need to have ATLEAST 3 blazing demo's.

If you think this is you than please send me an email including all the mentioned requirements above to: mirrrormixing @ aol . com with subject "UMG Korea Talent Search"

Looking for the next Chris Brown (DefJam)!

2008-12-28T13:22:00.000-08:00

DefJam in partnership with MirrorMixing Entertainment Group (MMEG) International (my company) is currently on the look out for the next Chris Brown. The requirements are: You need to be between the age of 13 and 16; of African American decent; an incredible singer and dancer, and you need to have ATLEAST 3 blazing demo's.

If you think this is you than please send me an email including all the mentioned requirements above to: mirrrormixing @ aol . com with subject "DefJam Talent Search".

Looking for Girl Group songs (major label)!

2008-12-28T13:18:00.000-08:00

The Grinehouse Publishing Works is currently on the look out for songs available for placed with a MAJOR LABEL GIRL GROUP with mass national AND international success.

REQUIREMENTS for submission are, that the song(s) needs to be of single material (meaning it is of HIT potential), and completely AVAILABLE and ORIGINAL (we DON'T accept current and former submitted material of any kind including material submitted to third party record labels and/ or artists, acts, groups, and bands). As a GUIDELINE for the song(s) think of groups with a R&B/ POP CROSSOVER APPEAL like "CHERISH" and "THE PUSSYCAT DOLLS"). We are only looking for the VERY BEST songs out there so if you have any doubters, please don't even try to, HOWEVER if you are CONFIDENT and think you have the PERFECT (HIT) SONG that suites the requirements as described above, than please send us an email including full details and the song as an MP3 attachment to: koen.grinehouse@gmail.com with subject "Girl Group Songs".

Thank you very much in advance!

K. Heldens aka The Music Surgeon

Phase Scrambling

2008-12-16T18:59:00.000-08:00

Intro
The following blog is a bit more advanced than the blogs I have posted in the past. In this blog I will explain how calculation of phase angle (phase difference) from time delay (time of arrival ITD) and frequency. And how this technique will help you mix-in your vocals better and smoother. The technique which uses this is called "Phase Scrambling" and is a pretty unknown technique in the world of recording and mixing music. However it is daily life procedure in radio and tv broadcasting.
I had promised to write an entire blog dedicated to phase to a few readers of my blog by the request of Michael Zick. The reason why I haven't been able to finish is the fact that phase is a very broad phenomena and on top of that it is very difficult and advanced to explain this in the 101 form. It includes matching levels, pseudo-stereo and the correct selection of a keyboard/ synth left or right side to use in the mix, haas effect, phase-cancellation, absolute polarity, and so on. Unfortunately I'm not able to publish a post about this phenomena, BUT it is a subject that will DEFINITELY be dealt with in my book "Painting the Picture of Music".

Because I felt the need to publish some thing related to phase, I thought it was best to publish a technique, which is so rare and unheard of by the entire community of mixing and recording engineers, to broaden, you my readers', horizon!Keep in mind that even this blog requires some knowledge of physics and math!

Connection Between Phase, Phase Angle, Frequency, and Time of Arrival
Coming back to the subject of this blog: "Phase Scrambling". First I want to explain a little about the word phase in general. Phase has a clear definition for pure traveling sine waves (ever tried to time-shift a duplicate channel?), but not for music signals. As mentioned in an earlier blog; all equalizers shift phase, unless it uses some "special" tricks. Phases are ALWAYS phase differences (you still following me?). Polarity reversal (pol-rev) is not a phase shift and therefore not recognized on the time axis (t).

If there is a phase shift or a phase delay of the phase angle in degrees it has to be specified between which pure signals (sine waves) it appears. For example: A phase shift can be between the stereo signals; left and right, between input and output signals (e.g. EQ), between the voltage and current, or even between sound pressure (p) and velocity (v) of the air particles.

By now I could have posted up the entire formula to calculate this. For those interested, please hit me up with an email on royalcollegeofsurgeonsinc @ gmail . com

So what has time delay to do with phase angle you might wonder?

Phase Angle
Lord Rayleigh came up with a theory back in 1907, known as the duplex theory. This theory contributes to understanding the procedure of natural hearing with humans. It is the very simple realization that the interaural time of arrival differences (ITD) are important at frequencies below 800 Hz as phase differences with the direction localization as ear signals, while at frequencies above 1,600 Hz only the interaural level differences (IDL) are effective. Between the ears the maximum delay amounts to 0.63 ms. The phase difference for individual frequencies can be calculated. This is one of the most important reasons why I set my stereo imager at around 1,088 Hz (which means every thing below this frequency remains untouched spatially, and every thing above will get expanded).

You can shift single pure frequencies (sine waves), but this is impossible for music programs (!!!).

Phase Shift

Conditions for distortion-free transmission taken from the book "Microphone Book - chapter 7" by manufacturer Schoeps.

While the demand for constant frequency (amplitude is clear, the linear phase needs rather explanation.There are engineers that expect the ideal phase as constant as the amplitude repsonse. However this is NOT TRUE! Initially, the phase begins at 0 degrees because the lowest frequencies end at 0 Hz (at DC). Note: There's no phase angle between DC voltages, and as you know DC delivers the cleanest signal possible.

In the course of giving a frequency a phase angle is without meaning, if the phase angle is only twice (2x) as large in the case of double frequency, and three times (3x) as large in triplicates, and so on.

To make this more understandable I will take the example of a digital delay in the following.
Let's take a signal and delay it by 1 ms, and recombine them at output into onesingle signal. The levels of both, the original and the delayed signals going in the mixer are equal, and the signal is set at, say 1 kHz (sine wave).

A sine wave of 1,500 Hz frequency (period T = 0.667 ms) and it's delay duplicate at 1 ms. The mixed signal will result in a signal with no amplitude, or complete cancellation of the signal (!!!).

The phase shift for any frequency with a delay of 1 ms. The diagonal line represents the increasing phase shift as a function of frequency. Note: Think of 540 degrees as being, effectively, the same as 180 degrees.

Another phenomena using phase shift is "comb filtering".

Time, Phase, Frequency, and Delay
Polarity and phase are often mistakenly seem as the same thing. They are not.

As a rule: polarity reversal (console phase switch) is NO phase shift of 180 degrees (time delay).
Polarity reversal (PolRev) is a term which is often confused with that of phase (phi) but there's no phase shift or time delay involved. Polarity reversal happens whenever we change the "sign" of the amplitude values of the signal. In the analogue world this can be done with an inverting amp, transformer, or the classic method; by switching the connections between hot and cold on one end of a balanced line cable. In the digital world this is done by changing all pulses to minuses and the other way round in a audio signal (data) stream.
Therefore the typical phase (phi) button is ONLY a polarity changer (!!!).

Conclusion
By knowing all the above, let's imagine a simple tool that can "eliminate" these problems by selective frequency and sweeps. Your audio signal will become more solid, and you will be able to tread it better and smoother. Wouldn't that be cool?

Waves has a plugin that is capable of doing this. Many of you have seen the plugin already countless times, and most likely have used it a zillion times before; the Waves PS22 Spread (TDM only). The PS22 Spread has a built in phase scrambler, but hardly any one knows about this.

Recently I have been working with this plugin as a phase scrambler and it shows impeccable results. My vocal is more solid, stays in the center image (most records the lead vocal moves a bit around) and my limiter hasn't have to work as hard as it used to. By placing the PS22 Spread first in chain and the L2 second. If you do this the other way round (limiter first, phase scrambler second; you'll get the opposite effect).

How Does a Phase Scrambler Operate?

Phase scrambler calculates the phase angles of an audio signal to reduce peak levels and accomplish a better (more solid and sustaining) RMS.

My Settings

Plugin: Waves PS22 Spread (TDM only)

Preset: Heldens Phase Scrambler

Width: 1
Rotation: 0
Spread: 0
LFspread: x1
Sweeps: between 2 to 22 to determine how much phase scrambling you want. My default is set to 8.
FCenter & FDensity: Can be set to adjust the frequency range to which the most scambling is applied.

Tip: The greatest degree of scrambling occuring in frequency ranges where the display graph say: Spread= 1.2 (most wiggles per octave)

Applications: Vocals or even entire mixdowns (before limiting! After limiting will have the opposite effect) to reduce peak levels.

Analyzing tip: To see how phase is being scrambled by looking at frequency, is to rest Spread to 2.1 (Set back to 0 for actual processing!)

--

If you guys wanna have the actual math (formula's) behind this please shoot me an email at royalcollegeofsurgeonsinc @ gmail . com and I will be more then happy to send you all formula's.

Signing off,

Yours Truly!

The Grinehouse Publishing Works

2008-12-10T18:14:00.000-08:00

The Grinehouse (Sony BMG Entertainment) USA, is a production powerhouse conglomerate of multi platinum songwriters and producers, and mixing engineers' Jimmy Douglas and The Music Surgeon (Koen Heldens) under direction of Michael L. Martin Jr. (CEO) and Dashawn White (CEO). In their short period of existance they have sold a total of over 20 million records and various number one and top ten positions in the American Billboard charts. They are one of the most sought after production houses in the world. The inhouse A&R is Kathy Hughes, Vice President of A&R at Sony BMG Entertainment USA.

Producers and songwriters please hit me up with your 3 best productions and/ or songs: koen.grinehouse@gmail.com

Good luck!

The Music Surgeon Koen "The Sound Butler" Heldens

M/S Compression over L/R Compression

2008-12-10T18:05:00.001-08:00

Intro
As many of you already know there's already a blog about M/S techniques during mixdown. Today I want to talk about M/S Compression which will be an extremely valuable tool once you master it. Mastering engineers make use of this but during the mixdown phase this can be an essential tool as well.

Pro's over Stereo (L/R) Compression
We are all familiar with what M/S does and where it stands for (Mid/ Side). M/S compression will give you another angle to how you will tread your stereo buss (mix buss). The M channel is the whole sum of the song, and the S channel represents the difference between left and right.

So M/S compression lets you compress and control the center, and sides of your stereo mix separately. This, all the sudden, allows you to bring up your vocals without affecting your instruments, bringing them back. Taming kick and snare on your overheads, or even emphasizing wide early reflections in the stereo field.

Scenario
A lot of times you've finished a mix, however sometimes the vocal gets slightly buried when the backing track (instruments) get loud. If you go on and try to compress the overall mix, or by the use of narrow band compression of the vocal's frequency range, you will notice that the compression will ruin the great sounding backing track (instruments). M/S compression will come in handy during this stage. M/S compression can help to isolate the center image (M channel). How? By only compressing the M channel, bring up the center when the signals get loud, or another option is to compress the M channel and expand the S channel. This way you help to control the vocalist and open up the specific band. By compressing the S channel only, anything panned center is unaffected and the compression only affects signals panned left and/ or right that are out of phase. Loud signals in these modes will cause a momentary reduction in level of the S channel and therefore resulting in a narrowing of image width. Another option is multi band M/S compression but I'm not going to touch that option today as it offers more possibilities. If you do work in analogue I strongly recommend to use a stereo compressor (which side-chains are linked, the reason for this is that you don't want any phase-shifts happening or an imaging change. In L/R compression it's not guaranteed absolutely zero, as analogue compressors will not 100% handle both channels absolutely equally and therefore some degree of image shift and phase shift might occur. However in M/S compression, any disparity between the channels will not result in any degree of image shift, but in a variation of the width of the stereo image which is less obtrusive than the degree of phase and image shift occurring in L/R compression. I personally would say it works way better than L/R compression and you will find that out for yourself once you've played around with it a few times.

How to achieve M/S Compression
Pass the M signal through one channel of a compressor and the S signal through another. Plugin wise you have to this in two stages.

Email me with all your remaining questions or your findings:
royalcollegeofsurgeonsinc @ gmail . com

Signing off,

Yours Truly!

Update: Long time...

2008-12-08T18:14:00.000-08:00

Just a quick note to let you guys know I haven't forgotten about you! I've been so busy and hectic ever since I got back. Due to family circumstances and work I haven't been able to share some time with you. HOWEVER, very soon I will start posting up some new blogs. Currently I'm in the studio with producer Drew Lane and will soon be back in the studio with Killa Kela to finish off his album.

One other thing, some of you already heard the rumour about me becomming 50% company partners with the famous The Grinehouse Publishing Works lead by two incredible multi platinum producers/ songwriters and Sony BMG Entertainment's VP of A&R Kathy Hughes. This rumour is true! Soon I will be posting more info on this. In the meanwhile If any of you guys want to submit tracks please email me on koen.grinehouse@gmail.com.

That's it for now I hope every thing is good with you'all.

Much musical love & blessings,

Yours truly!

Killa Kela in the house!

2008-11-18T08:49:00.000-08:00

Killa Kela, for those who are not familiar with his name, has appeared onstage with Pharrell, Justin Timberlake, Super Furry Animals, Prince, De La Soul, and Plan B. He is considered to be one of the best beatboxers in the world alongside Kenny Muhammad and Rahzel. Kela released the album "Elocution" in 2005, the follow up to his debut album "The Permanent Marker" in 2002. He is also known for a UK advert for "Fruit Pastilles" and regular appearances on TV shows such as "Big Brother's Little Brother".

Currently Kela is finishing his new studio album with legendary UK producer Martin Rushent best known for his countless hit records from Human League, The Stranglers, Altered Images, and Generation-X. Martin won a Brit Award for Best British Producer back in 1982 for Human League's album "Dare".

I had the pleasure to work on the first spin off (single) of Kela's new album.

UK Top 40 Act "Addictive" Mix Session

2008-11-13T13:07:00.000-08:00

Just a quick insight on me mixing the new single "Hot" by UK Top 40 Act Addictive. Music produced by IKONIX. Video shot by the studio owner Jason Ong (Jasonair Music Studio).

Antares Sells "T-Pain/ Cher" Effect

2008-11-09T13:18:00.001-08:00

First off all I want to thank my friend Stylo for sharing this information.

Intro
He hit me up through iChat today informing me that Anatares has made a "new" plugin named Antares Auto-Tune EFX. Which in basic terms, is just a simplified Auto-Tune with extreme tuning settings which can be manually switched between soft, medium, and extreme.

Antares Auto-Tune EVO vs. Antares Auto-Tune EFX
I remember loads of people in studios world wide were already making the joke that Antares should use T-Pain in their add campaigns to get more sales, or even sell a seperate plugin; which right now has happened. However, the very first one to do that T-Pain effect was on the track "Believe" by Cher. My main question is: Why would you want to buy a plugin like this? To further explain this, we all need tuning. Why buy tuning and tuning efx? Meanwhile both are already in one, which is the actually tuning (pitch correction) plugin. For me it's simple, just buy Antares Auto-Tune EVO.

So how does the T-Pain effect work?

In Automatic Detection Mode(not graphical)
C Major (or whatever root key the singer sings)
Retune-0
Tracking-67
Vibrato Section: All zero except for;
Amplitude-30
Formant-100
Pitch knob can be tweaked for more effect.
Voice input type: Alto/Tenor, or your choice.
Retune is usually always at 0, but the Tracking can be tweaked for different effects.
The Vibrato section is real important for the effect. All other vibrato settings except for those listed here are set to "0"(example- "rate"). Usually I have maxed out the Formant and Amplitude settings, but its track dependant.

NOTE: Auto Tune needs to be correcting the pitch to get good effect. Therefore I first apply Waves Tune on the vocal to get it sounding right before I apply the Auto-Tune effect. This way the effect will sound much smoother and solid.

Signing out,

Yours Truly

Links
Antares Auto-Tune EVO
Antares Auto-Tune EFX

Pro Tools Keyboard Shortcuts Added!

2008-11-08T22:45:00.000-08:00

I have added the Pro Tools 7 keyboard shortcuts in the menu on the right hand side under the header "Important Documents for Clients". The reason for this is to give students and people using Pro Tools but don't own it a chance to study the various keyboard shortcuts. Listed are both Mac OS.x and Windows XP keyboard shortcuts. For those who own Pro Tools, you can find the entire list (for the version you are working with) under the menu "Help".

Although the keyboard shortcuts are from version 7, they are still active in 8.

Good luck!

The Music Surgeon

"Snapshot" Automation in Pro Tools

2008-11-08T19:56:00.000-08:00

Today a very good question draw my attention when I was reading my emails. I got a question that came forward about an earlier post of mine, the "Starting the Mix" blog post of last month.

Intro
The question basically was about the way we mix a song in sections. The reason for this is quite simple; every song contains different phases, these phases are engaged to different types of emotional expressions. For example the verse might be more relaxed, meanwhile the prehook rises it's emotion and by the time the hook (chorus) hits, the song becomes loud and very emotional. We can't simply mix these phases in the same way we thread the verses, bridge, intro, outro's, and so on. Therefore we like to use the so called "snapshot" automation. This is available on both Pro Tools as the Solid State Logic, AMS Neve consoles among many other analogue and digital mixing consoles. To make this blog post more mainstream and understandable for most of the readers I will explain only how "snapshot" automation works in Pro Tools.

What is "Snapshot" Automation?
"Snapshot" in general is basically just the way of writting a static mix. Meaning that you will only write one setting, or series of settings with the same time stamp of the mixers setup across a section of your mix. To bring back the example given before, mixing in this way you will be able to make your mix suite the structure of the song. You can set up all the levels, mutes, voice keyboard, drum processing, and so on for each section (phase) of the song and write them as the foundation of your (final) mix.

So how does it work (Pro Tools HD)?

Select the selector tool to highlight acoss the section where you want to write your "snapshot" automation (e.g. chorus). NOTE: make sure you have selected all the tracks and parts you want to be included in the "snapshot" and that the selection is of the correct length.
Enable playback looping (ctrl + click on the transports "play" button). I'm always mixing in this mode.
Click Auto Suspend in the Automation Enable Window (command + 4 on numeric keypad).
You will now be able to loop around your selection (section) and set up your mix. When done press Stop.
Choose Write Automation to all Enabled Parameters (command + option + forward slash).
Take off Auto Suspend in the Automation Enable Window (command + 4 on numeric keypad).

Thats it!TIP: I always work in Relative Grid Mode while mixing. This mode allows you to move "regions" while retaining their position relative to the nearest beat. In practice this would, for example, mean that when you move a "region" who's starting point is 360 samples after the beat, can be moved without causing the "region" to automatically snap to the beat. Which would have been the case in Absolute Grid Mode.

Hope this little entry will help you guys get mixes going with various states of emotions!

Signing off,

Yours truly!

Good Morning Mister President!

2008-11-05T00:47:00.000-08:00

Thats how the UK's free news paper Metro opened today, and that's exactly how I'm feeling right now. I have to admit that during my time in the United States I have always been worried about where we, not only as country, but as world were going. This very moment I have an exciting feeling, a feeling of hope, hope of a brighter future for all of us, but most importently for our kids, the worlds future! A legendary moment when Obama got announced as the new president of the United States of America. The very first black president that will change the world in one way or the other.

This is not my normal music industry blog, however it does have a musical twist.

The past hours I was watching CNN closely from the K-West Hotel in central London. I was accompanied by a good friend Lyric and Beyonce's very own bass player the gorgeous and mad talented MC Divinity aka Divi Baby. It was an emotional moment for all of us. Meanwhile a lot of Beyonce's band members and Pink's band members needed to catch some sleep for todays gig at the MTV European Music Awards in Liverpool (3 hour drive from London). No one could get atleast some sleep, every one was way to excited to find out wheather Obama would win or not.

I'm so gratefull for MC Divinity aka Divi Baby to let us watch the entire race from start to finish with her on her Apple Mac Book. THANK YOU VERY MUCH FOR SHARING THIS LEGENDARY AND ONCE IN A LIFE TIME MOMENT IN HISTORY!!!! P.s. She's one of the most realest persons I have ever met. (Click picture to visit her Official MySpace page.)

I'm excited to hit back the USA (finally) this very month and reunite with my beautiful fiancee and her kids.

Change is here and is for good! Watch my words world!

Love you all and hope to catch up on some music techniques soon!

Yours Sincerely,

The Music Surgeon

M/S Techniques in Mixing; Mastering Legend Bob Katz Explains

2008-11-02T19:29:00.000-08:00

Bob Katz and I talk from time to time about mixing engineers delivering sessions to the mastering engineer and the various ways of work (Us vs them and vice versa).

Quite a while ago I send him an email concerning my earlier post (which can be found here) about M/S Techniques in mixing, but using it as an effect and how it affects the work of the mastering engineer, if it does at all!

Click here to find out what Bob replied.

When do I know I Over-compressed?

2008-10-28T18:08:00.000-07:00

Over-compression happens a lot with processing drums, even we as pro's and even pro's that have been mixing over a decade or longer still struggle with it from time to time. However there's a monitoring trick to hear whether your record has over-compression in the lower regions or if it's all fine.

The trick to hear if you are indead over-compressing your lower region material (drums) is that when you've been working for about, say an hour or two, on low volume (I hope you guys do cause else your records will sound to muddy!!!) and when you bring up the volume your drums should get louder in stages as well. Now if you are over-compressing this won't happen. Yes, the volume will get louder, but its more mid but not more low-end and punch. If this happens you have indead over-compressed your lower regions!

Hope this little trick will help you guys when mixing music!

Signing off,

Yours truly!

Starting the Mix

2008-10-21T19:33:00.000-07:00

Before I start mixing I make sure the client understands my way of working and the way they need to prepare the session. It's not much work just a little organising work so we understand what is what and where what is. However, even with the best we get Pro Tools Sessions that are massively disorganised. This is not a big deal, however it will cost the client an additional fee. This is the very first stage were my assistant comes in, s/he will prepare the Pro Tools Session. Cleaning things up, organising everything, put the different tracks to gether that belong together, assign tracks and channels in places that I'm used to and familiar with. That way I can find every thing back quickly and without using my brain. You can blind fold me on that! lol. Desk wise, the drums come up centre (left), the vocals come up centre (right). That way I can grab the fader(s) instinctively as explained before. I will name every thing in my session, from the busses, to the I/O's to the plug-ins. While all this happens I'm usually listening to the rough mix of the track which the producer includes in the package. This way I can get a general idea and feeling of the track, as ment by the producer.

Now that the preparing phase is done I'll make a rough balance of the song, by throwing up the faders one by one. Once this is done (usually happens within 20 mins) my assistant will print that rough mix. This way when I start building I can A/B between the rough mix I just made several minutes or hours ago and the one I'm working on this very minute. I also listen if I need to make additional overdubs, or need to move things around and the overal tuning of the instruments; which, by the way, is not an uncommon thing. I start by cleaning, polishing, and sculpting the kick drum, followed by the snare, which than again is followed by the lead vocal, add libs, backing vocals, and harmonies, followed by the bass (which after this phase goes on mute, as it contains to much energy I rather sculp it more when the entire track is standing so it fits in the best way possible). When these channels are sounding solid, I crank up the volume knob of the main monitors to go really loud and hurt my ears. If this sounds good I'll turn back the volume knob a lot and start working on the other elements of the mix. As long as my key elements of the songs (the rythmic section) is standig solid, I know my mix is going to be good.

At the end of the day when I heard some thing in the original rough mix of the producer that I find is missing or could be interesting to the track, I will incorporate it. Usually clients are surprised and satisfied by the difference in my mix approach and their rough mix.

[ENDED] INTERNSHIP OPPORTUNITY FOR CURRENT AND GRADUATED STUDENTS [ENDED]

2008-10-20T15:31:00.000-07:00

Multi Platinum and multi American Billboard number 1 mixing engineer is in London and currently looking for a qualified assistant (mix) engineer for a four day and four hour internship at a top notch commercial Solid State Logic mix room in North London. He is currently chart topping in the US with his work on "Womanizer" from Britney Spears. Latest credits include; Beyonce, Keri Hilson, and Timbaland.

The candidate should be available for four (4), 12 hour days, and four (4) additional hours following.

- Living in London or near the North London area
- Knowledge of Pro Tools HD is required
- Practical experience and knowledge of Solid State Logic 4000 and it's G+ computer are required
- Perfect attitude and hard worker
- Familiar with studio etiquettes
- Looking for a once in a lifetime experience
- Paid travel expenses
- Availabilty in the next two weeks (exact dates will folow asap)

*Note: a commercial camera crew will be on location to record a documentary of the client!

To apply please send your CV's and cover letters and course information (school, head lecturer, and college manager) to royalcollegeofsurgeonsinc @ gmail . com or call 07501-808-006.

RESPOND SOON!!!

UPDATE: INTERNSHIP IS CLOSED!!!! THANK YOU TO ALL STUDENTS WHO SUBMITTED THEIR COVER LETTERS AND RESUME'S!!!

Congratulations to Britney Spears!

2008-10-17T15:20:00.000-07:00

Congratulations to Britney Spears for reaching the Billboards number one position this week from position 96 last week (and all of that in her second week of its release!!!) with her latest single "Womanizer"!!! And of course a little bit of excitement from my side for working on the official remix of "Womanizer"!

Respect to Brandy Norwood!

2008-10-16T19:39:00.001-07:00

I want to dedicate this blog to Brandy Norwood. I've seen her work for quite a few months at The Boom Boom Room Recording Studio, Studio C in the spring of this year on her incredible album "Human". Unfortunately we missed out on actually getting to work with each other for various reasons.

I've heard the album coming together and was already excited to hear the end result. She worked with great talents in songwriting and production so it's no wonder that this album turns out great.

In my very honest opinion, both as a listener (consumer) and professional (mixing enigeer), I do think that Brandy Norwood is a very underrated singer and artist. Therefore please listen carefully to her new album and single(s)!

To come back at the album; A side from the single my other favourite track is "Fall" produced by Bryan-Michael Cox and written by British popstar Natasha Beddingfield, and Miss Norwood herself.

Check out the first spin off of the album, the single "Right Here (Departed)" produced by Rodney "Darkchild Jerkins and written by Rodney "Darkchild" Jerkins, and The Writing Camp members Evan "Kidd" Bogart, Victoria Horn, Erika Nuri, and David "DQ" Quiñones.

Brandy will be releasing her album called “Human” via Epic Records on November 11th,2008.

Signing off,

Yours truly!

Editing Beyonce, Britney Spears, Keri Hilson & Timbaland

2008-10-11T08:57:00.000-07:00

Yesterday was an interesting day. As I was in between sessions I got a last minute request to go and edit official radio edits (remixes) of the new singles from Beyonce, Britney Spears and Keri Hilson & Timbaland. After receiving the files I gave them a listen while aligning the tracks and listening to instructions about the specific songs and edits.

I had to fly some tracks around and insert some bars here and their, extend or shorten the intro or bridge, remove a chorus, play around with the songs arrangements and so on, as theres a well known east coast rapper (guess who) featured on these new versions. After all it was a very nice and quick gig.

For now I'd say look out for the following tracks on the radio. Go request them and give them some support!

Signing off,

The PT Editor, lol!

The Tracks:

Beyonce - "Single Ladies (Put a Ring on It)"
Columbia Records/Sony BMG Entertainment

Britney Spears - "Womanizer"

Jive Records/ Zomba Records

Keri Hilson (featuring Timbaland) - "Return the Favor"
Interscope Records/ Universal Music Group

Harmonics 101 ("Painting the Picture of Music" book excerpt!!!)

2008-10-08T20:30:00.001-07:00

This post actually represents another chapter from my forthcoming book "Painting the Picture of Music".

OK first my apologies for taking so long to post this up, but the matter of fact is that this subject still remains very difficult to explain to those without any music theoretical background.

However here it is and I hope I managed to get this subject a bit clear and more understandable to my readers. In case question still remain, and without a doubt I know there will be, please contact me and I'll try to explain them to you.

For now, enjoy!

The Music Surgeon for Royal College of Surgeons INK

Birth of the Harmonic
When a string vibrates the main pitch you hear is that of the whole string, back and forth (full cycle). This is called the fundamental (or firth harmonic).

Overtones or Harmonics?
The term overtone series is used to refer to a specific set of frequencies that appear above the musical tone. The term harmonic series is more precies and specifically refers to a set of numbers related by whole number ratios. Overtones and harmonics are often used to refer to the same thing by musicians, but in matter of fact are two completely different things. An example of an harmonic serie -the set of frequencies in Hertz (Hz) 1000, 2000, 3000, 4000, and so on. And so does 500, 1000, 1500, 2000, 2500, etc., make a harmonic series. The fundamental of the first example is 1000 Hz, for the second series this is 500 Hz. The other frequencies given in each example represent the so called harmonics or overtones.

Below I'll show you a little table showing you the difference between harmonics, and overtones.

(Click to enlarge)

As you can see, the first ovetone is actually the 2nd harmonic and the second overtone is the 3rd harmonic, and so on.

The simplest explanation for why such pattern of tones appears above the pitch one actually perceives may easily be found in a physical model of a plucked string as described in the chapter before. A monochord is simply a one fixed stringed device at both ends with a moveable bridge, and ofcourse a resonating body (timbre), and used back in the Middle Ages by sciencetists to investigate the relationship between string length and its relation to pitch (frequency). When the Monochord is plucked its string vibrates at a rate directly proportional to its length. However, it will also vibrate over fractional divisions of its length (1/2, 1/3, 1/4, and so on. Or proportional: 2x, 3x, 4x, and so on) producing higher frequencies. These frequencies are known as overtones.

The Missing Fundamental
Lets first start off to explain what Subjective Tones are to get a clear picture of what the missing fundamental actually is.

When two single frequency tones are present at the very same time they will, without a doubt, interfere with each other and produce a beat frequency. The beat frequency is equal to the difference between the frequencies of the two tones (an interesting artaffect is that when this happens in the mid frequency region the human ear will percieve this as a third tone.), called a subjective tone or difference tone. The difference tones are always present, but they can be made prominent by using two high, clear tones such as produced by the notes of a flute. By using two flutes you can produce a trio! This phenomenon can also be produced with one brass instrument and is refered to as mutliphonics, which I will not explain for now. Now, one important role of subjective tones is that of the missing fundamental.

We have learned that every pitch we perceive is based on the fundamental frequency of the sound rather than on any of the harmonics (overtones). The fundamental frequency has also got the greatest amplitude of all of the harmonics, but the reason we perceive only the pitch of the fundamental frequency is not due simply to its greater amplitude.

A missing fundamental is the missing fundamental frequency which higher frequencies refer to (harmonics/ overtones). A clear example has been given before. However low quality stereo speakers are not able to produce low frequencies, and so the 100 Hz (and below) frequency might not be reproduced by your speakers and will go missing. Nevertheless, a pitch corresponding to the fundamental will still be heard. Back in the day it was believed that this missing fundamental was replaced by distortions, introduced by the physics of our own ear. But experiments showed that when a noise was added, which frankly, would mask these distortions if they were present at all, but listeners would still hear a pitch corresponding to the missing fundamental. The reveal of this experiment shows and teached us that our brain processes the information in the harmonics (overtones) and calculates then what the missing fundamental is. The exact way our brain does this is still not completely clear scientists say its based on the timing of neural impulses in the auditory nerve.

This very same concept is used nowadays to reproduce the missing fundamental, based on the harmonics (overtones) of the signal. An illusion of bass will follow. By processings certain harmonics (overtones) selectively, a ricj bass effect can be produced which will be reproduced by the smallest speakers on the surface of planet earth. A company from Israel named Waves has succesfully produced a pattent product based on this principle named MaxxBass. RenaissanceBass is a simplified version of this.

Just Intervals?
Pythagoras plays an important role in the discovery of numerical relations between string length and musical intervals. Pitch produced by a string with a length of L/2 (length x2) sounds an octave higher than the pitch produced by a string of length L. This way it is easy for musicians to express the precise size of a musical interval. For example, there's an interval who's frequency ratio is 6:4 should be expressed using the ratio of 3:2. Get the idea?

So what about intervals than? Tonal interval names like a major third, minor second, or a perfect fifth are defined by their relative position within the major or minor scale. For example, the second and third tones of the harmonic series form a perfect fifth. The interval ratio for that ratio can now be determind by using frequency multiple numbers between the staves, 3:2. We can dig way deeper in to this by introducing scales and tuning systems but I want to leave it with this for now. Timbre is such a thing as well. MP3 compression makes use of this same principle but I will explain this more in my upcoming book "Painting the Picture of Music".

Legendary movie maker John Withney has made an exceptional example of the subject explained in this blog and is an animation coming straight out of his book "Digital Harmony". The link can be found in the External Links paragraph below.

External Links
John Whitney's Music Box
Waves RenaissanceBass
Waves MaxxBass
Waves Maxx Website

Waves Dorrough Meter Plugin

2008-10-07T15:29:00.000-07:00

Waves has finally as the first manufacturer of pro audio plugins and hardware, build a high quality audio loudness meter. Actually it's an emulation of the world known Dorrough 400-series Loudness Meter. I'm very familiar with them and got introduced to them during my days as a runner at a local broadcast station.

I can't stress how important proper metering is. Its part of my rule of having superb monitors, different pairs and types and brands of monitors, proper analyzing tools and so on. In this last category the Dorrough meter fits perfectly. In some way I'm surprised that waves is about to release (next month) their own emulated software version of the Dorrough meter but the biggest surprise is their price: $499,- for a meter, if you buy the original hardware version, which is $750 (however you need two of them to proper monitor your stereo). They target at the home and project studio but ofcourse also the pro studio. From what I've heard they want to introduce these kind of tools to the less financial fortunate studio owners (home and project) but with a price of $499,- for "just a meter" I find it a bit too expensive and if I would have the money I would definitely go and save a little more to buy the real deal. Hopefully at the end of the day they will drop down the price a bit.

Another issue I see coming with this meter are the different types. I don't know if they have build in the +14dB headroom version (400-A) and the +20dB headroom version (400-C), but most importantely information about why there are two version as this is critical for the type of music you are working on. Let me forward you to a great article from friend Bob Katz a seasoned mastering engineer in Florida: Click here to read his article.

Dorrough 400-A Loudness Meter

Above you see the Dorrough 400-A Series Loudness Meter. Now as you've read Bob Katz article, and I assum you did, you'll know now that 14dB of headroom is the reference for Pop music and thus the reference meter I'm using.

Dorrough 400-C Loudness Meter

Above you see the Dorrough 400-C Series Loudness Meter which is the "theatre" reference having 20dB of headroom.

Here you go! I hope this explained a bit about the infamous/ famous Dorrough meters and the Waves emulated version. In case you have any further questions about metering please shoot me an email on royalcollegeofsurgeonsinc @ gmail . com.

Signing off,

Yours truly.

Focusrite Liquid Mix HD: Dynamic Convolution Plugin

2008-10-07T13:34:00.000-07:00

As you know I'm proud to endorse Focusrite's Liquid Mix. During AES San Francisco this month Focusrite announced the release of another product within their Liquid range: the Liquid Mix HD.

The Liquid Mix HD is basically a stand alone plugin for Pro Tools HD. Instead of running it off your traditional Liquid Mix tactile control interface it will run off your Pro Tools cards DSP chips (as it's the plugin only) without any compromise in sound quality.

The Liquid Mix HD plugin includes the very same collection of classic vintage EQ and Compressor emulations as the traditional Liquid Mix range which are 40 Compressor, and 20 EQ emulations (and even more models online.)

When running an HD3 system, it allows you to run a total of 60 Liquid Mix HD Plugins (EQ and Compression) simultaniously within your Pro Tools session.

Guide Price is set around $499.-

Pro Tools 8

2008-10-04T15:33:00.000-07:00

Are you ready for the arrival of the new Pro Tools 8? I am! As we all know Digidesign got a bit left behind when it comes to their sequencer. OK I admit, they made some cool features available in the last update but still they were behind on other sequencers in terms of functionality and especially MIDI. MIDI has never been the main focus of Pro Tools as its a sequencer build for the pro recording industry and has always been highly audio driven and minded. However to keep up with the technology and the demand and need most users and studios are looking for nowadays they had to come up with a major update and here it is: Pro Tools 8. I won't enter in more keys on this keyboard but let you watch this beautiful demonstration video!

Pro Tools 8.x: New User Interface and Brief Overview from Scott Church on Vimeo.

I Love my Assistants

2008-09-29T17:01:00.000-07:00

Dear Readers,

Today I want to dedicate a blog about the hardest working people in the studio world. The assistans. Without them I won't be able to mix the records I do today. They are the driving force behind my mix sessions no matter what place and time on this planet earth. My assistant saves me a hell lot of time. He is the one that prepares all my mixing sessions, and trust me although I have posted up the guidelines and the majority of clients know about them; we still get a load of massively disorganised sessions, even with the best! My assistant cleans every thing up and organises everything, puts all the tracks that belong together next to each other, assigns tracks and channels in the way that is familiar to me so that I can start working straight away when I walk in. Elements will always end up on the same channels on the desk, and in the same way I will always have it. That way I can focus on mixing; mix blind, find my way in the dark basically. E.g. the kick drum always ends up at the left directly next to the drop frame (centre panel) of the SSL, followed by the snare, and so on. The Lead Vocal will always end up at the right directly next to the drop frame (centre section) followed by the add libs, harmonies, and so on. My start-up effects will always end up at the far end of the right side of the console, routed to aux/ echo 1, followed by 2 and so on; you get the idea. The drums and vocals always end up at the centre of the console as these are the most important things in a song. Knowing the placements allows me to instinctively grab the faders. I hand that very same principle for the Pro Tools sessions. My assistant and I do name plug-ins and tracks and so on as precisely as possible. I need to be in an artistic place.

Just once more a big thank you to all the assistants I had so far and the one(s) I'm currently using! You guys rock!

For any one currently following a course or study in audio engineer. Excel at your school and team up with a local studio, try and get an internship and hopefully one day you'll be able to do the mix yourselfs!

Yours Sincerely,

The Music Surgeon for Royal College of Surgeons, INK
P.s. SAE College Liverpool students watch out for my guest lecture at your college next month! For more info please ask your course' senior lecturer or your college manager.

RE: Loudness Level War

2008-09-27T14:31:00.001-07:00

Some thing kept me awake last night. I couldn't stop thinking of the fact who is actually responsible for the loudness level war. I remember posting a blog about this issue a few weeks ago. I explained how a lot of us, engineers, get pissed off and are pissed off about the fact that the consumer alias the listeners ears has been trained to accept the sound of distortion, and not the nice kind of distortion.

Anyways coming back to my thoughts of last night. The ones that are responsible for this ongoing loudness level war are we as engineers. Yes, we are. We invented the compressor. What was the reason for the compressor? It acts as a (auto) leveling device. I think thats where it all kind of started. I'm not blaming the compressor for what is going on lately but I do think the reason of the product and the mind of the engineer equals what is happening right now.

On another note I must say that in some way I kind of enjoy mixing this way. For certain songs this squishing really makes sense but for other songs not at all. I don't want to get a point of right and/ or wrong a cross but I rather would like to start a discussion about this.

Please comment on this blog with your findings and views.

Yours Sincerely,

The Music Surgeon

Sound Design Chapter 1: The Compressor

2008-09-26T10:40:00.001-07:00

One of my favourite subjects in mixing music is sound design. Most of you will link sound design with movies and games. However sound design is comon in mixing as well. Same tools, different approach. To start of my sound design chapters; the first piece of equipment is the compressor.

As many readers know I'm a huge fan of compressors. Most of you will use a compressor as a auto-leveling device, or simply a volume control. Instead I will use it to alter feel and affect of my instruments signal and performance. The compressor is commonley mistaken as the main processing within transient designers, this is wrong. Transient designers make us of different technology. I won't dig into this to avoid confusion.

The Technique
Back to the compressor; In the 1980's the invention of sound design with the use of a compressor saw the light of day. Mix engineers back then used compressors with a really fast attack, like my favourite kick drum compressor the dbx 160XT or later on the 165 series (no easy-over!!!). The settings were set extremely aggressive. You send your kick drum or snare signal throught it and put a gate post-compressor with the tightest setting possible. You'd end up with the attack of the sound only, phat and extremely aggressive (this technique often gets mistaken with how a transient designer actually works). You put this fader as the attack fader and blend it in with your other kick or snare fader. This way you would have full control over your kick or snare drum sound to get the degree of attack that you find most desirible. You can achieve this on any instrument you want and as you know from my multing/ splitting posts, it's what I do all the time. Simply choose the compressor for what you need. Aggressive compressors such as the dbx and the SSL type's so you can place the front end of the note to anywhere you want, or a Fairchild to get a more legato -sustaining effect (e.g. vocals) to gain a different feel for the track. That way you basically mix like a musician, or better you re-play the instrument as if you were the musician!

Attention to Detail
I still use the compressor -gate trick for my snare signal. On top of that I have the snare track duplicated and pitched (offline) down a fourth with a little reverb return (no direct sound) to add additional weight to the snare drum sound. However on the kick drum I go for another approach. I will compress extremely aggressive the kick drum, but will insert an EQ post-compressor with a high shelf at around 770 cycles and a large bell boost with about + 20 dB at around 10,000 cycles. This way I end up with attack as well. Both tricks work very well depending on the song type.

Play around with this trick and hit me back with your results!

Signing off,

Yours Truly!

P.s. still available SUPER REDUCED mixes for indie artists. Contact me asap if interested!

Mix Buss Compression

2008-09-24T09:47:00.000-07:00

Recently I have been receiving a lot of emails concering mix buss compression. The usual questions are about applying compression or not and when and where.

Mixing Through the Compressor
As most of my readers and my listeners know I mix-through-the-compressor from the very beginning. I first got introduced to this by legendary mixer Andy Wallace.

The reason for this, and especially with the SSL FX384 Compressor or Buss Compressor is that it will not interfere with my stereo image, as it would have when I inserted it at the end of the mix. But the biggest advantages over insterting a compressor at the end of the mix is that I will be able to create a very big sounding mix very quick. I don't need much automation which will save me a lot of time at the end of the day and on top of that my individual instruments don't need that much compression as it would the other way round. It will bring my mix together significant faster and my mix has way more punch already starting with only my kick, snare and bass. However keep in mind that after I set leveling for kick, snare, and bass, and lead vocal and backing vocals/ harmonies; I will mute the bass' channel simply cause it has too much energy to mix in the other elements properly. I will unmute the bass at nearly the end of the mix and sculp it in with the rest of the mix. In other words I adapt the bass to the mix and not the other way round, this makes life a lot easier.

This trick works extreme wonders for music that require a lot of low end and rythm or agressive music styles such as rock music.

Please note that if you are not familiar with mixing with a compressor on the mix buss at all, this requires some practice as the dynamics don't react in a way you expect them to do.

Ratio: 4:1 (on FX384), 2:1 (on 2254)
Attack: .10ms
Release: Set to 8th note or calculate BPM
Gain Reduction: 4 dBFS

Mix Buss Compressor at the End
Inserting a compressor at the end of the mix usually makes your stereo image a bit smaller than you'd have before it. You'll be able to create a bigger sound however you need to back and make a few changes. This technique requires a lot more work than the one I use.

Ratio: 2:1 (on FX384), 1:5 (on 2254)
Attack: .10ms
Release: Set to 8th note or calculate BPM
Gain Reduction: 2 dBFS

No Mix Buss Compression
Some mix engineers use this technique. You won't expect what I'm about to say:
This technique can truly sound AMAZING! Yes, I just said it. But why don't I use it than? Very good question and the answer lies in time. This technique requires an awful lot of time which nowadays I can't afford unfortunately. Maybe when I start working inhouse for a produce it's possible, or a big paying client, but for now it's not unfortunately.

This technique will create a very open and airy sound and will work very well on slow jams nad pop ballads. It will have more dynamics and more depth and subtlety which would have been lost when put through a compressor.

My favourite compressors for mix buss compression are:

SSL FX384 (Located next to the computer screen)

Zoom In of the FX384 (XLogic Series Module)

AMS Neve 2254 (x 2) (Located top right)

Zoom In of the 2254's

I hope this will answer all your question when it comes to mix buss compression. Please contact me if you have any questions left about this subject.

Signing off,

Yours Truly!

What's Wrong With My Frequency Analyzer?

2008-09-22T18:24:00.000-07:00

Although I push reading a frequency analyzer a lot, in this blog I will explain that you always need to rely on your ears and the reason why.

Introduction to Harmonics in Electronics
The typical definition of a harmonic is a sinusoidal waveform, or in short sine wave. Most manuals will refer back to this as a clean or pure signal. However in music it's a known fact that such thing doesn't exist and that's the reason why a guitar sounds like a guitar and so on. To get really down to harmonics in music we need to take into acount various other subjects such as the music temperaments etc. I will save you this.

The simplest way to explain harmonics is to start of with your every day electronics. Electric generators try to produce electric power where the voltage waveform only has one frequency associated with it. That frequency is called the fundamental frequency. Readers in the United States of America will know that this is 60 Hz and readers in Europe will know this is 50 Hz for them.

The frequency of the harmonics series is dependable on the fundamental frequency. The second harmonic for a 60 Hz system is 120Hz (2 x 60), for 50 Hz this is 100 Hz (2 x 50). 300 Hz will be the fifth harmonic in a 60 Hz system (5 x 60) or the sixth harmonic (6 x 50) in a 50 Hz system. Got it?

How Does a Frequency Analyzer Work?
To make things harder; I'm always pushing assistants to read a frequency analyzer when working, I'm talking about a FFT - Fast Fourier Transform. But there is one problem, to represent/ calculate a complicated waveform in a microprocessor and computer based system it requires mathematical steps and these are quite difficult. Therefore it will not always show exactly the right frequencies as it should. To give you an example of this: If a waveform exist out of 60 Hz and 200 Hz, the FFT can't directly see the 200 Hz as it only knows equal steps: 60, 120, 180, 240 and so on. These are normally called bins. So what happens with my 200 Hz then? Good question! Your 200 Hz signal will appear partially in the 180 Hz bin, and partially in the 240 Hz bin. These inbetween harmonics are simply called interharmonics. There is also a special category for interharmonics which are frequency values less than the fundamental frequency value. These are called sub-harmonics.

Conclusion
Because it doesn't represent exactly where your area of problem or timbre lies doesn't mean analyzers are useless. As in matter of fact to show you where the problem area or the sounds timbre is, it will take you a little listen and tweak on the EQ to find it exactly. Now you understand how electronical devices represent sound and frequency on a little display, you'll no longer give a strange look at your analyzer or pull your hair and think that the analyzer is not working properly.

Hopefully this helped you a bit. I try later this week or next month to write a 101 blog about Harmonic Series (Overtones) in music.

Signing off,

Yours Truly.

Preparing Your Sessions For Mixing

2008-09-20T18:43:00.000-07:00

Just a quick blog for all my current and future clients.

On the right side you can find the permanent link to the NARAS Grammy Foundation Producers & Engineers Wing Pro Tools Guidelines. Please read them, understand them, but most important use them. These guidelines are made in conjunction with legendary engineers and producers such as Charles Dye, Roger Nichols, Bob Ludwig, Tony Maserati, Mick Guzauski, and Erick Schilling to name just a few. The copy of this document comes straight from Charles Dye's personal website.

Here's the link

These guidelines will enable me to work faster and eliminate a lot of confusion and technical or pre-mix problems. In basic words, I will be able to start and mix your sessions straight away.

Another thing I'd like to inform you about is, if you work in other DAW platforms other than logic or pro tools please be advised to consilidate all your audio tracks with the duration of the song so I can line them up in the sequencer and every thing will play in time and I can start mixing straight away.

Thank you all for reading and I hope to work with you soon!

Yours Truly.

The Art of Picking the Right Drum Samples

2008-09-19T14:12:00.000-07:00

Todays post is some thing I find very important for all producers of all levels. But we only experience it with aspiring producers. Please watch the video posted below before you continue to finish reading this blog post.

First off all I don't want to go ahead and dis-credit my man Kayne West. He's an awsum producer and I have loads of respect for him. However this video shows one of the most important factors in producing music. With my background as a producer of dance music a few years back, I can't stress how important it is to pick the right drum samples for your song.

Picking the Right Drum Samples
Loads of times when I mix tracks for aspiring producers I have to deal with wrongly sampled and/ or picked drum samples. Samples including to much crackle and noise or even worse samples with nothing to it. In short samples we hardly can work with or do any thing with it at all. Sonically choosing the wrong samples does affect your final product in a major way sonically. You can hear Mr. West talk about it:

"Although the song is hot, the drums ain't right." - Kayne West about Stronger.

On top of that he added that because the drums aren't right, the kick drum doesn't hit in the clubs.

The main reason why that happens is perfectly explained by Timbaland in the very same video example above:

"You can't compress some thing that isn't there." - Timbaland about the kick drum in Stronger.

Please keep these two quotes in mind at all time. Write it down and live it, is basically what I want to tell you.

The lesson for today: Every time you produce a track go and find clean samples with body.

Layering Your Drums
Another thing I can't stress how important layering your drums is. Timbaland shows that more then clearly in the exampled video above.
A lot of tracks I receive for mixing from seasoned producers like Brian Michael Cox, Rodney "Darkchild" Jerkins, and others have three kicks, three snares. The reason for this is that some samples lack frequency respons, or you want to accentrate a certain frequency region such as the low end or attack. Therefore layering your drum sounds to make up for this and support your overal drum sound will make your beat sound much and much stronger. Also make sure when you use EQ during this stage that it is way more better to cut then boost.

The lesson on this for today: Beat making starts with your drums, so you have to make sure they’re tight before building on top of them.

Editing Your Samples
Adding to these issues it is also important to edit your samples when building your beat. Make sure every single samples starts from the start of the sample. There's nothing more annoying and lacking in "punch" than tracks with a beat where the samples aren't tight. Note: I'm not talking about quantizing or it's vallues, especially not the fantastic rythmic quantization groove of the Akai MPC's.
Another thing with editing your drum samples I want to address is to pitch them to the right root key. We need to tune them a lot of time and if you have got the luxury of an engineer during your production sessions, it isn't a problem but for those who work outside of their bedroom studios and can't effort that luxury, please listen and re-tune your samples to the right root key that will save you at the end of the day a lot of money and your demo's will sound sonically way better, more pleasant, and tighter.

I hope I have helped you in some way by posting this blog. Please be adviced if you are about to let me mix a session or you have questions in general please contact me via email on royalcollegeofsurgeonsinc @ gmail . com or via iChat/ AIM screenname mirrormixing.

Signing off,

Yours sincerely,

Waves SSL G-Master Buss Compressor - Stereo Compression

2008-09-16T13:47:00.000-07:00

Dear All,

I just realized that I had forgotten to post this one up. As I mentioned this blog in some recent posts. My apologies and go ahead and read the full indepth info on this subject.

Enjoy!

The Music Surgeon

When I used to work in external studios with legacy SSL boards like the G+ or J series, one of the very first things I did when starting out a mix was routing every thing to the Buss Compressor or patching out the Buss Compressor to use it as a 'external' dynamics signal processor on my drum group. A lot of students and aspiring mixing engineers have asked me the same question about stereo compression over and over again. How to stereo compress without ending up with a tiny mix? The trick is to reverse engineer the whole progress of stereo compression. Start with the stereo compressor from the very beginning instead of ending up with the stereo compressor. That way you eliminate the stereo image problems accruing when putting the stereo compressor at the end of the processing chain and still maintain phat, punchy and big sounding drums or mixes. This trick is named "mixing through the compressor".
How does it work?
Start out by opening your mix session and insert the Waves SSL G-Master Buss Compressor on either your Master Buss Fader or on the Group Buss that you would like to stereo compress (e.g. Drums). The threshold is set at such a point so as to control the higher level elements of the final mix. By decreasing this threshold, more compression takes place which can clearly be heard and seen on the VU Meter of the Waves SSL G-Master Buss Compressor. The overall reduction of the mix is around 3dB. For drums the reduction is 6dB on the loudest transients and around 3dB on the overall signal.

The ratio is set to its lowest setting of 2:1. At this setting the compressor has a very soft knee characteristic, meaning simply that even the low portions of the signal are affected by the gain reduction circuitry of the compressor, but less then the high level parts of the signal. For drums I like to set a ratio of 4:1 to make them sound more 'tighter' and 'punchier'.
The attack time is set to 10ms, which is fairly long. This allows the initially attack of the instruments to come through before the compressor kicks in. If you reduce the attack time, you will hear that your mix is going to sound more and more squashed as the compressor kicks in quicker. With the attack time set to 10ms you allow percussive transients to 'breathe'.
The release is set to 600ms, which again is fairly long but it allows the compressor to recover smooth. Set the release time shorter and you will notice that your mix start to 'pump'. The make up gain is set to make up for the loss in loudness during the compression.The Analog button is pressed to give the drums/ mix that nice analogue saturation.

*Keep in mind that the release time of the compressor is BPM dependent!

Limited Time Offer!

2008-09-15T21:09:00.001-07:00

For all the support you guys have given me I present:

(Click image to enlarge)

Frequency Represents Pitch: A Technical Journey to Music Theory

2008-09-14T13:06:00.000-07:00

A lot of times I see students, interns, and assistants grab the EQ without actually knowing what they are reaching for, both technical as music theoretical. This has been some sort of a problem and issue the past years I'm working as a professional mix engineer.

There is an important factor involed for schools offering audio engineering courses. I truly believe that if you want to make it big in this game you need to have a certain level and understanding off music theory. As in matter of fact, play and master an instrument. I hardly know any tracking or mixing engineer that doesn't play an instrument. Me myself I play a little keys and had music theory training at the age of six. It may sound as a cliche but a lot of engineers come from a band background, that very same applies to my back ground however I started out as a semi-pro music composer (dance).

To come back to the issue of frequencies; frequencies do represent a musical pitch, a note that some one played with his or her musical instrument. The first picture I want to share with every one is that of the frequency range and representation of an 88 keys piano.

Please download the image and studie it. Print it out and use it during your sessions. It will make a night to day difference.

Another image I'd like to share with you is the very same 88-key piano, but now with the fundamental information of the most common instruments related to it.

Again, studie this one. Download the image and print it. Hang it on your studio control room's wall. This will make the big difference in your mixes today and those of tomorrow and in the future.

For all those who want to go and attend an audio engineering course and don't have music theoretical training, please go and get you music theory 101 diploma before continuing persuing your dream to become an audio engineer.

Tips on high rated and successful schools providing an audio engineering course are:
Berklee College of Music, Boston MA*
Fullsail University, Miami FL

*Provides also summer courses

I hope this little blog will help you establish better sounding records.

Signing off,

Yours Truly.

From Motown to New York: Parallel Compression

2008-09-13T21:08:00.000-07:00

My mind says I should have posted this blog way earlier than I do right now.
Various public and private postings and lectures have and are been given about the subject of parallel compression.

Parallel compression came into my life when I was working as an assistant engineer back in the day via a good friend of mine from North Hollywood, engineer and mixer Richard Furch (Tribase Productions/ emixing.com) which I thank a lot for this. Later on in my career when I got introduced and met master mixer Dave Pensado I got introduced to a whole other way of parallel compression, that of multing or better known as splitting. This technique opened my world into a way better and controlled sound and overal mix.

As many of you know I have been talking about multing for quite a while in my blog posts but never really explained were this very technique came from and how to apply it.
Well the day has come that I will go and write a little blog entry about this and well to start with the history of parallel compression or known to others as New York compression.

History of Parallel Compression: The Motown Era
Back in the 1950's and 1960's records where generally mixed with the vocals a lot louder than the backing track. They usually put a lot of reverb on the backing track to make it blend more in to the background with the lead vocal dry and upfront. A good example for this is Frank Sinatra and Elvis Presly.

Than there was this up and comming record label named Motown who started to make a big buzz in the scene with a lot of talented people on board and a lot of creative rules to brake, their engineer had a different view and approach to traditional mixing. Their vision was to make catchy and dancable records, rythm was the key.

In 1970 Motown invented the Exciter Compression technique which was the very first type of parallel compression. A man named Lawrence Horn came up with a brilliant idea and, in my opinion has to be seen as the invetor of parallel compression. He splitted the lead vocal onto two different channels on his analogue console. The first channel he added some light EQ and some reverberation. On the second channel he compressed the hell out of the vocal signal and added a lot of high end EQ (around 5,000 cycles and up). Now he had two channels of lead vocals; one natural with EQ and reverberation, and one heavily compressed with tons of high end EQ. He than blended the second channel with the original channel just enough so it gave the lead vocal excitement.

The end result of this technique was that he ended up with a natural sounding vocal, clear and audible at all times with a sparkly and bright touch.

Motown was innovative back in the day and still remains a key factor in how we mix records in todays recording industry. As in matter of fact a lot of rock and pop records are mixed using the very same techniques as Motown once invented and used on their countless hit records.

The technque explained above works well on every single instrument but with a different approach as many readers will know from my earlier postings.

So, Where Does New York Compression Comes From Then?
Yes, I already thought this question might popped up in your head just about now and I don't blame you for wondering it.

So how does the term New York Compression got mixed up in this?

Back in the early-mid1990's a group of mixing engineers came out with a refreshing sound in America. They all came out of the great city of New York. One of these pioneers is legendary mix engineer Tony Maserati.
What they did wasn't that revolutionairy and new, as in matter of fact 30/ 40 years earlier another mix engineer by the name of Lawrence Horn was already doing this.

Not quite exactly. What the group of New York mixers did was taking Lawrence Horn's technique to whole other level. The New York sound is best known for their in your face instruments. Every thing upfront, meanwhile back in the Motown era the backing track and its instruments remaind blended in reverb on the background. This was not the case with the New York sound-ing mixes.

New York Compression comes from parallel compressing your drum group or buss. It works exactly the same as the Motown Exciter Compression but without the big chunk of EQ setting.

Intention of Parallel Compression
Parallel Compression's intention is to maintain the performance characteristics of an instrument or sound source, but with way more and better stabalized dynamics. The compressor settings are set in a way that causes the compresses to pump or breath. Blending both signals will result in a more human than static sounding compressor.

Multing (or Splitting)
The Parallel Compression trick is the very same trick we use on our mults (or splits). However we use it to gain more control over our audio program as explained in earlier posts.

I advice you to go and play around with this trick and let me know about your experiences and findings.

Signing off,

Yours truly.

Larger Than Life: Stereo Widening

2008-09-12T13:15:00.000-07:00

Stereo Widening tools are all over the place and often abused in the wrongest sound of the word by many producers and aspiring mix engineers. It's a subtle but for the sweet spot an incredible effect. When done right it makes your mix sound like a radio played song, or better a radio ready song (however I still advice to get it mastered). I use it on every single mix whatsoever. My mixes won't do justice without this a cross the stereo buss.

The History of Stereophonic Sound
The entire concept of recreating stereophonic sound was invented in the 1930s after various experiments by Dr. Harvey Fletcher of Harvard University. He had a dummy and cut off it's ears and replaced them with microphones. He then connected the microphones to his headphone, left mic for left side and vice-versa. During the experiment he walked around and noticed that when approaching the dummy he could hear exaclty what his position was towards the dummy. When speaking and walking he could hear his voice changing from one side to the other. This all is part of the beautiful subject binaural hearing. But to completely understand how our ears work please visit the MyDr. website.

To get back to our subject. The link between recreation of stereophonic sound and binaural hearing is time difference (remember Haas effect in my 3 Dimensional Mixing blog?). The longer the time gab between two sound sources, the more the listener percieves that the sound source moves to the side.

A better explanation lies more likely in the natural world, the animal world. Animals make use of a system called echolocation to determine the direction of sound. They send out a series of high pitched, ultrasonic sounds, and listens to objects reflections in its path. The same applies to blind people they listen to echo's bouncing off objects, and than will be able to create a mental image of the room.

What is a Stereo Widener?
A Stereo Widener exist in both the analogue, as a hardware device, and the digital, as a software plugin, world. Famous examples are the Behringer Virtualizer Pro, SPL Stereo Vitalizer, DuY Wide, and the Waves S1 Imager and S1 Shuffler, to name but a few. Its a tool that enables you to create a much wider stereo image.

Stereo Widening/ Imaging tools serve the purpose of manipulating (increasing, or decreasing ) your music's stereo image. They make use of your music's centre (mid= M) and side (left and right= S) information. By manipulating the relationship of the side signal (S) and the centre signal (M): M = (L + R)(1 / 2);S = (L - R)(1 / 2). A positive part of the side signal (S) is than fed into the left channel, and another part (with it's phase inverted- 180 degrees) to the right channel. Whenever your mix at the outcome sounds phasy, please go back in to your mix and revision it's mono compatibility as you've create some phase problems pre-stereo widening.

You can also re-create this phenomena, without extrapolating a centre (M) and signal from your left and right (S) by simply add the left signal (slightly attenuated and phase inverted- 180 degrees) into the right channel and vice-versa. You may add a little bit of delay (20 - 40ms) to the inverted signal before mixing it back with your original output. Please advice that adding a slight reverberation effect will make this effect sound more natural.

More About Stereo Widener Tools
The more intelligent stereo windening tools give you control over your low frequency spectrum and your high frequency spectrum. The reason for this is that low frequency sounds such as kick drum and bass sound best when placed in the centre (middle) of a music mix. Besides that fact, they are equally driven by both speakers. Not only that, they also contain a lot of energy and therefore having these sounds coming from the centre will make the speakers work "less" hard.
The kick and bass are the foundation of a mix. Placing them slightly off centre will make this appearing very obviously to the listener and will in most of the cases sound very odd to the listener. In psychological means, placing kick and bass in the centre of your mix is reassuring to the listener. Therefore some stereo widening tools divide the frequency spectrum in two parts, low frequency and high frequency, and give you control over each.

Getting More Mono Before Stereo Widening?!
This may seem a bit odd but after reading this part you'll understand why this is.

A lot of producers and aspiring mix engineers pan their material hard left and hard right. For drums this will sound very ackward as a drum kit isn't as big as the entire stage, unless you want to have a very wide sounding drum kit. Besides that, if your mix is very busy it will be a lot harder to place sounds around your drum kit and therefore it will be better to keep your overheads mono and in centre, or pan them slightly off centre. The band Radiohead is a classic example of having a mono drum sound as well as dance and club music. However dance and club music usually have a differenct approach and reasoning for this. The producer and mix engineer like to have their kick and bass sound equally throughout the entire club, left to centre to right. Therefore they like to keep their drum and bass sound mono instead of having parts of the club not enjoying every single sound which the drum kit is made out off.

The very same usually applies for vocals, especially lead vocals, unless they are doubled. When your vocals are doubles, usually background vocals and ad libs, its desirable to pan them hard left and hard right. The very same applies for rythmic guitars but not for synths! Yes you read it right now for synths. If you receive synth tracks in stereo, simply get rid of one side as the other side usually is just a delayed side or chorused side. The importance of mixing is to listen if some thing is really stereo and not pseudo-stereo! It will not only save you tracks but also gives you more control over mixes sound sources.

Keep in mind: "Bigger is not always better!"

How I Use My Stereo Widening Tools
To start off with my favourite one, Wide by Spanish company DuY (in my opinion) the most natural and usually is best suited for the job.

My Settings for DuY Wide:
Input is set to - 5dB as I will use Waves L3 afterwards.
Frequency is set to 1,088 cycles which is the in your face frequency.
Boost is set to 0% depending on the loss of low end frequency information.
Amount is set to 30% for a more 3 dimensional sound.

My Settings for Waves S1 Shuffler:
Input is set to -5dB as I will use Waves L3 afterwards.
Width is set to 2.00 to achieve a very 3 dimensional sound.
Shuffling is set between 1 and 4% depending on the loss of frequency.
Frequency is set to 1088 the in your face frequency.
Bass Trim is set between -1, 0, to +1 depending on the loss of low end frequency information.
Polarity and Channel Swap is set to default.
Asymmetry is set to 0.
Rotation is set to 0.

I don't use Waves S1 Imager as it won't give me enough control over my stereo image.

NOTE One thing to keep in mind, if you do use compression on your stereo buss and you do use the famous SSL Stereo Buss Compressor (FX384), please use it pre-mixing and not post-mixing as it will tend to make your stereo image appear and sound much smaller. For more info on this please refer back to my earlier post this year about the SSL Stereo Buss Compressor.

I hope this has made the subject of stereo imaging and widening more clear and more approchable for you.

If questions please hit me up with an email on royalcollegeofsurgeonsinc @ gmail . com

Signing off in London,

Yours truly.

Moment of Thought: My Favourite Pieces of Gear

2008-09-12T11:58:00.000-07:00

As a moment of thought, I thought I'd post a little list of my favourite pieces of gear.

Console:
Solid State Logic 9000J SuperAnalogue Series Console

Monitors:
Auratone 5C Radio Reference or Any laptop 3" speakers
Yamaha NS10 Nearfields
Barefoot Audio MicroMain 27 Midfields or PMC TB2S-A Nearfields
George Augsperger Augspergers

EQ's:
AMS Neve 1073DPA Stereo Mic Pre (as EQ)
API 550A Discrete 3-Band EQ
Avalon AD2055 Dual Mono Pure Class A Parametric Music Equalizer
GML 8200+ Parametric EQ
Focusrite ISA 115HD Parametric EQ
Solid State Logic XLogic X-Rack Channel EQ Module (set to G EQ characteristics) or an old SSL 4000 G series console channel strip

Compressor/ Limiters:
dbx 160XT Mono Compressor/ Limiter (no Easy-Over)
Solid State Logic FX384 Buss Compressor
AMS Neve 2254 Mono Limiter/ Compressor
Urei Teletronix LA2A Mono Limiter
Orban Optimod FM8100A Multiband FM Audio Processor/ Stereo Generator

Plugins:
Waves Diamond Bundle
Waves Studio Classics
DuY Global Bundle

Sequencers:
Digidesign Protools HD as industry standard and audio editing
Apple Logic Pro 7 for it's routing options

Stereo Buss Processing:

R&B and Pop records
AMS Neve 2254 Mono Limiter/ Compressor (dual) and Focusrite ISA 115HD Parametric EQ (for that rich sparkly sound)

Harder records:
SSL FX384 Buss Compressor and GML 8200+ Parametric EQ (for that typical SSL sound)

Let me know what you guys use/ like most!

Signing off,

Yours Truly

Liquid Mix > Koen Heldens - Hip-Hop Mix Engineer

2008-09-07T19:21:00.000-07:00

Hip-Hop Mix engineer Koen Heldens speaks to Rob Jones:
http://www.focusrite.com/featured_artists/artist_videos/liquid_mix/1/

Channel Side-Chaining

2008-09-03T09:01:00.000-07:00

Channel Side-Chaining, what does it mean and what does it do over "normal" Side-Chaining. Or even better, "What is Side-Chaining in general"?

Let me start with introducing those who are unknown with the phenomena of Side-Chaining, to retain basic knowledge about it. For those who are already familiar with the basic concept of Side-Chaining, please skip this paragraph.

The Concept
You will find side chain inputs on dynamic processors such as Compressors, Expanders, Gates, and Limiters. However you won't find them on EQ's, but why is that?
The EQ is used to color and or filter the instruments timbre (How it exactly works is described in the blog about "Low-End Phase Shift" located earlier in this months blog entry list). However in basic technical concept, the way a compressor or an EQ work is fairly the same and pretty autonomous. Wait a minute right here, the compressor has a side chain input, and (remember) the EQ hasn't. This means that the compressor can base its operation on a different signal, but how? Let me give you and example to show you how it works.

Imagine you have got a bassline that hits every down beat of the track, and therefore will hit the very same time as your kick drum hits. Due to both signals low-end energy content you need to make space for each and another. How better to achieve this then using the so called "ducking" technique which works as following: When the bassline hits every single time the kick hits simultaniously during the track, the compressor will reduce the bassline's gain by the amount given on the compressor.

So how do you set some thing like this up?
What you do is send the kick drums signal to the "external key" input of the compressor so it willbase it's baisc operation on a completely different signal, which in this case is that of the kick drum. Now switch the compressor to trigger from the external key. Now the compressor will gain reduce the bassline's signal every time the kick and the bassline hit at the same time.

This technique is also very often, and commonly used by radio DJ's to talk over the track that's playing. Get the idea?

The side-chain opens on compressors, gates, limiters, and expandors have many different other uses as well, technical and creative.

The above example is one of the many side-chaining techniques I use daily. Another one is the one I want to talk about today is the so called "channel side-chaining". Well what does that mean and do?

Channel Side-Chaining
Channel Side-Chaining is a techniques most commonly used on rock music or any music recorded with "real" instruments. Another most common name for this is the Dynamic EQ. The EQ of the channel strip contols the channel's compressors base operation. I can let the compressor frequency dependent distress, de-ess, compress, expand, and even gate any signal on that specific channel.

What I do on electric guitar for example is creating a de-esser (EQ and Filters to Side-Chain) on the channel itself and set the frequency usually around 5,000 cycles to remove the harsh sound of the electric guitar.

Another example is drums, I usually High Pass or Dynamic-EQ (EQ and Filters to Side-Chain) the snare's and overheads to create more space and rest within the overal drum sound.

The Creative Approach
A creative approach to Side-Chaining is to set unusual triggers. Some examples are triggering the reverb tail of the snare with your hihats or side chain your shakers with the kick drum. On vocal effects I usually side chain the dry vocals with the wet vocal effects, it works as follows: The trick is to place the compressor after the delay or reverb. You set-up another stereo Aux Input track and send the dry vocal group track to the Aux Input track, and do another send from the dry vocal group track to the compressor's external side-chain input. It's a great trick but take in account it will only work on dedicated individual effects rather than global effects.

For those who produce dance music will be familiar with the ducking effect as far as the main lead/ pad triggered with the songs main kick drum.

But the biggest tip I can give you is: "Go creazy there are no rules!"

I hope this blog will clearify a bit more about the subject of side-chaining.

Email me for more info or questions on royalcollegeofsurgeonsinc @ gmail . com

Level Loudness War

2008-08-28T08:14:00.000-07:00

The tittle of this blog will be familiar with loads of people reading this and I'm also confident that a lot of people will find this subject -or what I rather call an annoying issue (that has been going on for the past, almost, two decades.) one to discus about.

After the 80's the level war started to come up and ever since all the records we hear have less dynamics and right now they even have fewer than ever before.

We have accepted the sound of distortion, and by distortion I don't mean the pleasant harmonic distortion type I love to use during mixing. Our ears and brain have been exposed to distortion for such a long time it has become an every day phenomena, which is very unfortunate. We have made the consumer, the listener of music lazy. After all they are the ones that control the volume knob and can adjust it to their listening needs... not us the professionals!

I won't spill more words and time on this, but instead want you as the reader to check out this beautiful video made by Matt Mayfield about the Level Loudness War.

And please have a look on the dedicated website about the Level Loudness War.

Please discus your views and this via the comments section of this blog.

Signing off,

Yours Truly

Solid Low End Trick

2008-08-18T06:55:00.000-07:00

Dear All,

It has been a little while since I posted a new blog.

Today I want to talk about solid low end and how to achieve this.

Imagine this: you have a kick drum, sounds nice has body but still there's no significant detailed low end. It is not sounding solid. For us humans to identify a kick drum our ear searches for the low-end (usually around 50 cycles to 65 cycles) content and it's attack (usually around 3,500 cycles to 5,000 cycles). Usually I mult my kick drum or kick drums. I split them into three parts which the original only gets EQ'd by the board EQ from the Solid State Logic 9080J, the second one gets compressed the sh*t out of it via a DBX compressor or Waves Renaissance Compressor and API 550A's as the EQ's and becomes my parallel compressed fader. The third one gets the same threadment but a bit more extreme with Waves C-1 Compressor and a high pass shelf at around 770 cycles and becomes the attack fader. A side from all this I also use Waves Renaissance Bass for its harmonical contribution and Waves TransX to redesign/ shape the overal kick sound so it suites the song better.

Now the kick drum is sounding phat and round however it's low end is a bit undefined. What I do then is hit up the Solid State Logic 9080J board's OSC (oscilator) (You can also achieve this in Pro Tools itself by using Pro Tools' own Signal Generator. Put it on 50 cycles and -6dBFS using a Sine Wave). I route this one to one of the empty channels, put it on 50 cycles and side-chain Gate it with the original kick drum signal, so that whenever the original kick drum hits -the oscilator's Gate opens. When the kick drum passed -the oscilator's Gate closes. You get the idea?

I always use 50 cycles, however I re-tune it with Waves Renaissance Bass and make it sound more phat and solid. I EQ it afterwards and hit it up with Waves TransX to blend it better with the overal kick sound. If you would solo the fader now you'll hear that it sounds more like a kick drum than an annoying hum sound.

Blend the OSC fader with the other kick drum faders and you've created a solid sounding kick drum.

Signing off,

Yours truly.

Low-End Phase Shift

2008-08-04T20:02:00.000-07:00

Today I would like the talk about how EQ's actually work, how they apply it to a signal, but in particularly on the low-end regions.

A lot of people always ask me what I find so wonderful about my API Lunchbox loaded with 550A modules. They only have fixed values...? Yes, they do and being a little off with the frequencies, especially in "multing mode" (splitting of channels) it's a desirable affect. Not only that but they also have almost no phase shift, as one of the few EQ's available, when boosting the low-end regions of my kick drum information.

When working in the analogue environment ever thought of the way processors process your audio programs? Well EQ is one of those things when applied to the low-end it can cause phase shifts. In other words it will screw with your signals time. You'll get an time offset when phasing occurs and this can either contribute to your split or it will affect it in a very nasty way and the overall signal will lose it's power. The other question which walks parallel with this one is: Do you have the ability to hear time offset between the tweeter and woofer? Try this out at home or at the studio....

Analogue Circuitries vs. Digital Binaries
In the analogue days all EQ's were made out of electric circuitry with capacitors and inductors. What these components over time do is shift the phase of the AC signals passing through them. When, after capacitor or inductor, you combine both signals -the original and the EQ'd version (phase shifted), there will be a different frequency response. Please advice your Frequency Analyzer for visual representation when you can't hear it. When the original cycle is rising, the EQ'd one on the other hand is still on it's way or is already falling which will result in cancellation of certain frequencies along the way which in audio terms will cause a non-flat frequency response.

Analogue EQ's work by adding the both signals together like described above which will cause an unfortunate and in some cases unwanted affect, especially when talking low-end information as these contain the largest cycles.

You might think: Why didn't they ever made a solution for this phase shifting affect? Well, simply because without it they would never work!

Digital EQ's minimize the behavior that causes the unfortunate affect of analogue EQ's. However they do work completely different. They replace the capacitors and inductors that cause phase shifts with digital tap delays on digital delay lines which make use of the sample and hold principle. In order to create a digital EQ from a digital delay line you tap into one of the buffers and feed an amount back into it's input. A change of polarity is also a way to achieve the same effect of boost or cut. However even this, as you can read, will create a phase shift!

At the end of the day any and every EQ on the surface of the planet earth causes phase shifts on the audio programs it's applied to. The only way to ovoid phase shifts within EQ's is to use special made ones, the so called linear-phase EQ's.

Now you know what EQ's do and what kind of affect tracks they leave on your audio, it is wise to decide and figure out what EQ you want to apply on

what application a side from the color you may like from a particular EQ.

My choice for the API 550A's on the low-end has two specific reasons, one mention all the way in the beginning of the blog post -little to no phase shift, and the second one: it's beautiful coloration.

Low-Level Compression vs. High-Level Compression

2008-08-02T19:34:00.000-07:00

Ever since the manufacturer Waves Audio, Ltd. came out with their, in my opinion, revolutionary product MaxxVolume this year I’ve been using it on a lot of live played instruments, but in particular vocals. Usually I needed to do a gain ride on my vocals on every different pass a different type of gain ride. Then compress it with about -3dBFS and then knock of about -7dBFS to -9dBFS on a limiter to achieve a solid sounding vocal in the mix. But thanks to their invention of MaxxVolume it saves me tons of time to do other, creative, things in my mixes. No more gain-rides, and no more limiting!

In a recent conversation with Waves co-owner Gilad Keren I thanked him and gave him my congrats on the invention of MaxxVolume. What a great and time saving plugin and concept I just addressed in the paragraph before. Unfortunately the theory behind the concept is very unknown to many, many engineers. I was surprised by the little amount of students, aspiring engineers, and engineers that actually do know, and do understand about low-level compression versus high-level compression and what it actually does.

MaxxVolume got advertised as an ‘automatic gain level rider’. This is in fact true, however I need you to take notice that there’s certain knowledge involved to get this beast to do what it’s supposed to do and what’s it made for.

Let’s go back in time and explain the basic theory behind the traditional compressor as we know it which uses high-level compression.

High-Level Compression
In basic terms, this is already an automatic volume control. A High-Level compressor reduces the gain, another term often used is level, of the specific audio signal if its level exceeds the set threshold. The amount of gain-reduction is determined by the set ratio. (E.g. with the ratio of 2:1 a 2dBFS input will be reduced to 1dBFS over the set threshold on the output).

The Attack settings means how fast the compressor will kick in, get to work. The Release setting means how fast the compressor will recover back to unity gain.

Remember it now?

Low-Level Compression
Low-Level Compression can be explained in simple terms as it does exactly the opposite as a High-Level Compressor. A Low-Level Compressor will upwards compress the gain when the level falls below the set threshold resulting in an increase in gain or in other words a distressor. The dynamic range will be compressed by pushing up the low level information but leaving the high level information untouched.

This is how a Low-Level Compression curve looks like:

So what does Low-Level Compression vs. High-Level Compression do?
Softer passages will become louder, and louder passages will become softer!

Go and try out the Waves MaxxVolume and post me replies on your findings with this beautiful plugin! http://www.waves.com/content.aspx?id=369

Signing off,

Yours truly!

PAINTING THE PICTURE OF MUSIC

2008-08-01T13:54:00.000-07:00

DEMO BOOK EXCERPT !!!

INTRODUCTION
In the 70’s the records that came out were really 3D sounding and had a lot of colors to it, towards the 80’s and 90’s the loudness war levels began and thereby also ended, unfortunately, the “dynamic” period. So what defines a 3D mix, or better said, how can you achieve a 3D mix? I will give it a go with what in my theory is a 3D mix however it’s way to complicated to address it correctly. There are various issues involved but I honestly think with my theory you will be able to achieve a good sounding 3D mix. No boring pan-potted mono!

Keep in mind that I will only address these various basic issues briefly. Of course there’s a lot more to it but I do think these are the basic principles and they will help you to create a better sounding mix at the end of the day.

For a full read please stand by until the book is officially published.

2-CHANNEL VS. MONOPHONIC
Our binaural hearing apparatus is largely responsible for the perception of depth.

Lets therefore first address the recording process and monophonic versus 2-channel recordings. I want to address the subject of masking. Masking is a term used to describe the following: A louder sound will tend to cover a softer sound, especially when they are in the same frequency area’s. Lets say the loudest signal is the direct sound of the sound source and its reverberation is the softer signal. When the initial direct sound will cease the reverberation is perceived.

Lets assume for this example that you’re listening to a pop concert in large sports centre such as the LA Staples complex in California. The monitors standing in front of the state will generate you the direct sound, and there will be sound bouncing of the walls which will give you the ambience/ reverberation, the 3D sound. The last one we will perceive diffusively from all directions. In the mono-phonic recording days we would recreate this phenomena by the use of artificial reverb units. However the reverberation would come out of the same sound source as it’s direct sound, and if you did use the same amount of reverberation as you would in a stereo environment, the direct sound could mash away and drown in it’s on reverberation and become inaudible. Nowadays we do have the beautiful way of recording instruments 2-channel. And we can register the example above, however this has to be done correctly, phase, timbre and so on all play an important role to this.

2 DIMENSIONAL
“DEPTH & DIMENSION” THE FRONT TO BACK PRINCIPLE

“HAAS” EFFECT
Another issue we need to address is the subject of the “Haas” effect and can be used as a tool to ovoid directional masking as I explained in the paragraph before. The Haas effect says that the human brain isn’t able to process and conclude when two signals under approximately 40ms are out of time from each other and therefore should be perceived as an echo. Instead it will fuse the both, direct and delayed, signals together as one and only a loudness enhancement will be perceived. Even when they each come from a different sound source! However our brain will continue to recognize binaurally the location of the original sound source. Therefore it will enhance the original sound source without confusing it’s directionality.

To become more proficient about using the “Haas” effect in mixing, Google and study the various papers on the various fusion effects at different delay and amplitude ratio’s.

EARLY REFLECTIONS
How to distinguish ambience, room and hall reverberation which is uncorrelated with that of correlated reverberation (echo’s) from nearby wall and floor of the direct sound source?

Actually in the question already hides the answer. The shorter the echo’s (reverberation) the clearer the direct sound source will be and thus the reverberation enhances the direct sound source and therefore is a correlated reverberation. The signal will also sound closer to the listener.

REVERB
Adding a lot of echo’s, especially long types of it, will cause the signal to drawn in it’s own reverberation and therefore it will sound like a room, hall, etc. an ambience. The signal will sound further away, a large distance from the listener and it’s sound source reverberation therefore will be an uncorrelated one.

Reverberation contributes to the perception of distance.

HIGH FREQUENCY ABSORPTION
Another important issue we have to address is that of high frequencies.

Have you ever noticed that when something fades and starts to disappear it’s high end information starts to cease? And vice versa, it’s high end information can become annoyingly loud? The sound source will be upfront and in your face?

This example contributes to the sense of distance as well. This phenomena has to do with the absorption qualities of air. In understandable words: When the distance of a sound source increases, the apparent high frequency response is reduced.

Within the mixdown process we can simulate and recreate the same effect by rolling off high end information to make a sound source appear further away from the listener. Or the other way around, by adding tons of high end information to make it appear in front of the listener. In his/ her face.

This is a good practice together with reverberation to place your instruments within the mix on a specific spot and space.

The above issues and subjects is what I call the “front to back” principle.

However something really important to address here is the rule of room radius, better known to most of you as the critical distance. Which in basic terms means that for every doubling of distance from a direct sound source you get a 6dB drop in SPL (sound pressure level). However this rule only applies on free-field, anechoic chamber, or at the top of a pole outdoors. So what happens in a real room?

ROOM RADIUS AKA CRITICAL DISTANCE
Combine the information you just read before this subject about reverberation and keep this in mind while I’m going to explain what room radius, or better known as critical distance, is.

Of course there must most be a middle distance where the direct sound source’ level is equal to the reflected (reverberation) sound pressure level (SPL). This distance is known as the room radius or critical distance.

You can go all technical about this subject and calculate this by using Equation 3.59. Note: the value is dependent on the volume of the room (V) and the reverberation time (RT60).

3 DIMENSIONAL
“UP & DOWN” THE PRINCIPLE OF HEIGHT

HEIGHT
The “front to back” principle also applies to “up and down” or in better words “height”. To achieve up, we need more reverberation. To achieve down on the ground we need less reverberation. However you have to interlock frequency absorption with this as well. The higher up you move the MORE high frequencies you perceive. This is because the high frequencies tend to rise (radiate) upwards rather than forward.

This issue does add some complexity to the problem, as I explained before that high frequencies get less with distance. However beyond the room radius, or critical distance point we do not perceive any significant changes in high frequency response when height is changed.

All the above factors affect the depth picture.

PANNING
Panning is another tool in 3D mixing.

It is a much easier principle as it will only place a direct sound source in the left, centre, or right regions of your mix’ spectrum. However panning direct sound sources to the extreme’s of your mix’ spectrum, hard left or hard right will cause serious problems when done before hand (pre-) stereo widener (imager). The stereo imager will cause to place those direct sound sources “outside” of the speakers which will cause to make your mix sound extremely big and very un-natural. I do advice and highly recommend to not go all the way to the extreme’s of the spectrum with your panner. Avoid hard left and hard r

ight, and do revision your mix balances pan-wise after you applied stereo imaging (widening) to avoid a to big sounding, un-natural, and phasy mix. Also always check for mono-compatibility at all times. If for some reason in mono any of your mix’ critical instruments disappear you have a problem and need to re-check panning, delay, reverb an

d all of those other things and adjust them by bringing them down, -less effect.

The pan issue is what I call the “left to right” principle.

AMPLITUDE (LEVEL)
Another simple factor in mixing is that of amplitude or better said level. This does exactly the opposite of panning. Amplitude, or level is another tool to achieve the “front to back” principle of distance. However it does walk hand in hand with the issu

e of reverberation and high frequency absorption.

The softer the direct sound source the further away, the more distance. The louder the direct sound source lesser distance and closer to the listener. This is also a tool to make things appear small in the mix.

Some thing more I like to address is the use of transient designers. I’m a big fan of these and together with mix engineer colleague Dave Pensado we use it on everything. It uses additive synthesis, and basic ADSR (Attack, Decay, Sustain, Release) as most synth users and programmers are familiar with. I won’t address this m

ore but if you want to know more about this subject please shoot me an email on royalcollegeofsurgeons @ gmail . com.

TRANSIENT DESIGN
Transients appear closer to the listener when they are the steepest and sharpest. It will contain a lot of high end energy and therefore, as you’ve just read before, will appear closer to the listener. Less transients therefore will appear further away, greater distance.

With the war levels going on for about the past two decades, I find transient design a very useful tool to get more “punch” on direct sound sources as these will get reduced and less due to the level wars. However it will also be much easier to place the direct sound sources sonically in the mix.

Color Coding or Colour Coding?

2008-07-28T14:25:00.000-07:00

I just had this major feeling of addressing the use of colors during mixdown (or tracking in case you are a recording engineer).

I know a few mixing engineers that don't wanna use the Pro Tools mix window but instead they make use of color coding. I do the same thing. I always have my little tool kit with me including a metal case with 24 different colored pencils in there, a roll of painters tape to put on the desk above and underneath the large faders -or to place on top of certain buttons on the console, a normal standard grey colored pencil, an eraser, different colored markers, normal black marker, and if I'm lucky enough different colored painters tape (which I highly recommend over different colored pencils and markers). Thats it for my tool kit.

Within Pro Tools I'd love to color my tracks, drums red vocals blue and so on. I do this cause it makes it way easier to find your sound sources back especially when you just hear some thing you wanna fine tune or adjust quickly. This very same principle I apply to the console. I roll down my tape, get my Pro Tools outputs running from the drop frame (center section) to the sides (most played sounds come first, least played sounds will end up all the way to the ends of the console). I write down the names of the tracks in colors with my pencils. The very same color for Pro Tools and that very same color on the very same track on the console. If I do am lucky enough to got a hold of colored painters tape, which is -and I have to acknowledge it, very hard to find, I will use that on top of my regular white colored painters tape and write down on it with my regular grey pencil. But the coloring principle is the same as with my colored pencils, but with the different colored tapes. Once done the mix is done and printed my assistant will write over the pencil text with the indelible colored or regular black markers, depending if I used the colored tape or the colored pencils. This way its extremely convenient and quick to find back the sound sources you are looking for while mixing or while on the recall for the mix.

I advice you to try this out if you haven't done this yet. Unfortunately I'm unable to post pictures of this yet but with one of my next sessions I hope to be able to take some nice pictures and post them in my blog.

Signing off,

Yours Truly ;)

What We Do It For

2008-07-25T20:43:00.000-07:00

We don't do it for the fame, we do it cause we love to do it. We love to have no social life, we love to be in the studio 24/7, we love to work an a single instrument for hours, we love it as a hobby that has become a profession. And I tell you, from that moment it is not a joke, clients pay serious amounts of money. I know I will let the client leave with a good mix no matter what. I was sick as a dog few months back but I still went to the studio and finished the mix. This might have not been my best work to date but the client at least left with a high standard mix, though I was puking in the trashcan every twenty minutes or so. I do what I get paid for.

Anyways to cut to the point. Today I received a phone call from an artist (rapper) telling me that he A/B'd the radio single I mixed for him. He told me that it sounded like a complete new track in a very good way! Those kind of comments keep me going, clients all satisfied and happy with the final result. He even hired me to do his entire album! Thats some thing I just wanted to share with all of you. I'm happy when I get phone calls like that a side from the mass emails I'm receiving lately from my publication of "Mixing the Magic of Music".

I don't do it for the money, trust me but I do have to get some bread and food every day and make a living....

Thanks again to every one and please keep on emailing me questions, thank you notes, and things to discus about.

royalcollegeofsurgeonsinc @ gmail . com

Signing off,

Yours Truly!

Free Waves Artist Presets from The Music Surgeon

2008-07-25T09:59:00.000-07:00

We're in the middle of the summer and because of all this heat and good weather I thought I'd surprise you all with my custom artists presets for Waves plugins for FREE, yes you read it right for FREE.

Windows XP/Vista users click HERE
Mac OSx users click HERE

The bundle has 36 multi-presets for Waves Diamond Bundle (or Platinum and Transform Bundles) and Studio Classics (or API, SSL 4000, and V-Series Bundles).

Enjoy and hit me back with your experiences!

Signing off,

Yours Truly

Probably My Biggest Processing Secret

2008-07-24T21:31:00.000-07:00

Yes, I thought I'd finally publish this to the public official.

One, and probably my biggest secret in processing my drum group and vocals, and even some times a cross my master stereo buss is an old vintage product produced by a manufacturer which goes by the name of Orban. Most recording and mixing engineers aren't familiar with their products and name or even in some cases don't even know about its existence.

Orban comes from the broadcasting world of pro audio equipment. A serious amount of world-class radio stations have got their equipment and uses it.
I come from a background in radio. That's were I started out my career as 'engineer', though it was as runner I did keep my eyes and ears open for what the studio tech-heads were talking about and that's were I got introduced to the sweet sound of Orban.

Its not uncommon that equipment from other area's of the audio industry find their way into the control room of a recording/ mix studio. One great example is the compressor. I spare you the details, but if you insist in knowing it please Google it.

Lets get back to Orban, and the specific model I'm talking about. Below are the pictures of my model the Optimod-FM 8100x made for FM radio transmition.

1980
The multiband OPTIMOD-FM 8100A was introduced, later to become the most popular FM audio processor/stereo generator ever made. The 8100A is the first Orban FM product to use the patented distortion-cancelling clipper.

I leave the little plated door on the right open at all times to adjust the parameters for the specific audio program I running through this phat sounding tool. I use AES/EBU out via a AES/EBU to XLR adaptor from Neutrik. This will cost you about $50,- but you'll be able to route it back to your patch bay and make it available as patch option.

So what's this Orban Optimod-FM 8100x tech specs?
2 Band AGC (Dual threshold wide band/narrow band AGC: A radio receiver incorporates an AGC system having an upper wide band threshold and a narrow band threshold. Both thresholds are made temperature independent. The narrow band threshold is related to the circuit response to tuning meter voltage and the wide band threshold is related to the tuner output prior to i-f selectivity. When these responses are logically anded, an AGC response having the desirable characteristics of both narrow band and wide band systems is achieved. An IC form of AGC circuit is disclosed. Noncoherent-coherent A.C. coupled base band AGC receiver: The present invention describes a phase shift key receiver or demodulator having an A.C. couple base band automatic gain control. A pair of detectors for the automatic gain control are A.C. coupled to the output of a pair of linear analog multipliers for the purpose of eliminating DC offset signals and for minimizing thermal noise at the input of the automatic gain control circuit. The outputs of the pair of detectors connected in the data detecting branch and the carrier tracking branch of the PLL are connected to a input of the summing circuit whose output is connected to the automatic gain control loop filter. The output of the filter supplies the scaling signal employed as the scaling input to the linear analog multipliers.), Compression, Limiting Variable Pre- Emphasis Limiter, Smart Clipper, FCS (Frequency-contoured side chain) Overshoot Compensator (Multiband cross-coupled compressor with overshoot protection circuit. A multiband compressor suitable for compressing the dynamic range of an audio signal in a broadcasting system is disclosed. The frequency band with the highest predictable energy is selected as a master band with the energy level in this band used to control the gain in the other (slave) bands. When the energy in a slave band exceeds a predetermined level, additional control of the signal level in that band occurs. The described compressor has the advantages of both the wideband compressors and multiband compressors without the disadvantages of these compressors. In addition to the compressor, a unique overshoot protection circuit is described.), Distortion Canceling Circuit (Distortion cancel circuit in FM stereo: There is disclosed a distortion cancel circuit in FM stereo receiver, which circuit synthesizes a canceling signal mainly including the distortion component by performing the arithmetic operation for an FM demodulation signal and adjusts the level of this synthesized canceling signal and thereafter subtractively combining it with the FM demodulation signal. This circuit enables the elimination of harmonic distortion which appears in the stereo MPX signal output due to the band characteristic of an intermediate frequency amplifier.), Linear Non Overshoot Compensated Low pass Filters (In terms of broad inclusion, the filter overshoot control circuitry of the invention comprises a first clipping circuit which functions to establish input signal amplitude limits, cooperating with a phase-lag network which displaces high frequency signal components with respect to low frequency signal components, and a second clipping circuit to re-establish signal amplitude limits, with the output therefrom being fed into a comparator circuit which recovers those signal components lost in the second clipping process, and which feeds such signal components into combining circuitry to re-introduce recovered, clipped signal components from the second clipping operation back into the primary information signal, but in opposite phase.), Very Stable Analog PLL Stereo Coder (Stereo imager.).

The best thing about this box is that it sounds extremely warm. And I mean desirable warm! Whatever you do, you almost can't go wrong with this box.... tweak a bit of the settings and you'll be blown away.
I can type for hours about this box but you've got to hear it for your self. I have it in my collection and it will stay there next to my Gates equipment for years to come. I highly suggest every one to try and find one on the second hand market (please don't go and buy the new digital versions!) and try it out yourselves. You'll be stunned and blown away by how this beautiful gold 'beast' sounds!

Signing off,

Yours truly!

Time Consistent Dynamic Design

2008-07-23T13:55:00.000-07:00

The header of this blog may sound a bit difficult to understand but when you’ve read through my entire blog it will turn on your creative ‘light’ and you’ll respond with an ‘ahhhh’;). I will go ahead and make you understand with what I mean by “time consistent dynamic design.”

OK what I wanna talk about is some thing we do a lot during mixdown; splitting up our instruments. Today I’d love to talk about how I split up my drums. I will explain exactly how I thread my kick and my snare, the drum group and the additional hi-hats.

It is the ‘punch’ I’m looking for, and by punch we mean ‘distortion’ or in other words ‘overtones’ which are harmonics. For drums I’ve always found that the Solid State Logic FX384 compressor, or better know as the G-Master Buss Stereo Compressor can achieve this best. But I do mix ‘through’ this compressor instead of inserting it afterwards. The reason why is that I can start working on the overall sound of my drums, and thus the overall sound and the amount of punch I want to have I have a thing for using stereo buss compressors on individual instruments instead of the entire 2 track (master buss). I do overload, or like we say it ‘squish’ the compressors output. NOT the gain reduction (!!!). On my drum group I use a ratio of 4:1. It is pretty extreme, but the FX384 compressor from Solid State Logic sounds equal to any other compressor with that setting, that’s set to a 2:1 ratio. An attack of 10ms so that compressor will let the initial attack’s through, and a release of 600ms depending on the speed of the song. You can calculate the release time. You do the same trick as you would calculate the delay time. The formula for this is 60,000 divided by the BPM (beats per minute) speed of your song. This way the compressor’s release recovers quick enough before the next attack of the kick drum kicks in. If however your song is written in a non 4/4^th straight beat please adjust the time by ear until it you achieve the same result. I do highly recommend to listen by ear at all times (!!!).

Now that we’ve set up the drum group aux input track and the Solid State Logic FX384 stereo compressor including its starting point settings, we continue and create another aux input track. This track will function as our kick drum group track. Yes you read it right ‘kick drum group’ track. Why I call this my ‘kick drum group’ is because I will split my kick drum twice and send all three (3) channels to the aux input track of my kick drum group’s input via the channels outputs, no sends but straight I/O’s (!!!). Now that we’ve done that I will insert the Solid State Logic FX384 stereo compressor on my kick drum group track and set the attack to 10ms and the release to an equal ms time. This way it lets the initial attack of my kick drum go through and the release will recover quick enough for a ‘phat’ in your stomach punch and a in-your-face-slap-of-attack kick drum sound.

After the Solid State Logic FX384 compressor I insert the beautiful Waves Renaissance Bass followed by Trans-X Multi and Wide plugins. Yes, this is going to be extreme transient design but it’s an 808 kick after all, and all three plugins are set to bypass only the compressor is active (!!!).

On the original kick drum I use only EQ and only the Solid State Logic board EQ, on the first split I compress the sh*t out of it first and then add EQ I do like the API 550A’s best for this and it has to do with its characteristic sound –distortion aka harmonics, this fader I call the MUD channel which allows me to blend it in with original and form a human compressor. The second split and third channel I compress even way more extreme and post EQ with an extreme 18dB bell-type boost at 10kHz, and roll off all low end content as for this channel I only want to capture the attack of the kick drum. I do all equalization post-compressor as a compressor tends to lift your low-end energy. So filtering, cutting, and shelving afterwards is a way better solution to avoid these problems.

Now with all three channels threaded; I blend them together to form a nice and solid kick drum sound. When done I go back to my kick drum group track, I will adjust the threshold and output to a better sound, and afterwards enable Renaissance Bass, I set the frequency to my kick drums fundamental tone which in this case was 64Hz. The amount of harmonics are set to -2dB and the output is adjusted to -3dBFS. The kick sounds already more solid but I do want to have some more design control therefore I enable Trans-X Multi, now I want to go in and achieve more low end and high end attack. It still needs some more overall attack, this is where I enable Trans-X Wide and turn up my transient some more until I achieve the desired amount of attack for this kick drum.

I do the same for the snare, I do like SSL equalization best for this but I will only use the Trans-X Wide in many cases as I don’t want to over design the snare’s top end.

For hi-hats I usual don’t use any of this, only some equalization and compression when needed.

As all the aux group tracks are already I/O’d to the drum group aux track I will go to this track and adjust it’s compressor’s threshold and output to a desired amount.

I split damn nearly all the instruments in a mix because I will have more control over them and can make them sound exactly the way I have in my head.

So to come back to this blogs header: what do I mean with “Time Consistent Dynamic Design”? A compressor is time consistent, you decide when the attack kicks in and when the release recovers back to unity gain and all of this happens over time –time consistent. The same applies for transient design. The only static part in all of this is the EQ however compression, transient design, and EQ are all dynamic design tools, but when we combine compression and transient design we’re working over time and therefore I’d like to call it “Time Consistent Dynamic Design”.

Signing off,

Yours truly.

WORLD RESPONS ON "THE MAGIC OF MIXING MUSIC"

2008-07-21T16:20:00.000-07:00

Recently we have made up the balance of my blog "The Magic of Mixing Music" about mixing down UK R&B sensation Keisha White's latest single "Wrong and Right". The response and reposts of this blog have made forums and printed paper all over the world some examples are USA, United Kingdom, Holland, Germany and Sweden. The response from aspiring audio engineers, students, colleagues and assistants was overwhelming and therefore I'm about to create an entire book with interactive DVD about mixing music for Hal Leonard book publishing!

I would like to thank every one reading my blogs and for all those generous emails I have received from you all! Thanks again!

Yours sincerely,

The Music Surgeon

THE MAGIC OF MIXING MUSIC

2008-07-17T21:27:00.000-07:00

There has always been this mysterious picture about mixing music. We all know what happens during a recording session. It has been seen various times on TV and engineers and artists talk very openly about this part of the (pre)production process. However when we switch over to the mixing phase of (pre)production it all because very mysterious. We have heard certain things and to an extend we know what goes down during a mixing session in a very basic sense. Mixing engineers appear to close up when asked, instead of opening up, sharing, and starting discussion about varies techniques. Luckily this is slightly changing nowadays and information, thanks to the internet becomes more accessible. Thanks to websites and magazines like Sound on Sound and Waves.com. The last one in particular is offering free plugin presets recently for Waves product owners.

Why be secretive about what we do, what tricks of the trade we are using and applying when we are, nine out of ten times, 're-inventing the wheel' anyway. Besides the music industry is a billion dollar industry with plenty of artists and producers out their in need for mixing engineers. I can't be there to do it all, neither can be any colleague of mine. Why not hire interns with potential and loads of talent and above all the right attitude who eventually will, and be able to, fulfill a position within your company? Get more clients, give back to the community, and on top of that earning money doing this. To me this only seems like a Win -Win situation, but I can be wrong.

Recently I was over at Larrabee Studios on Lankershim Blvd. in Studio City, a block away from where I live. I met up with colleague and master-mixer Dave Pensado discussing these very same issues. Dave, who runs two rooms within the Larrabee facility (B & C) simultaneously, occupied by his second engineer and his assistant, and an additional set of runners, has that structure I'm talking about. I also need to acknowledge that Dave taught me a lot of tricks among another guy -dear friend and colleague Richard Furch (emixing.com/ tribase productions) and I have always been supported to the fact of sharing, discussing, and publishing mixing techniques. However I never like my mixes anyway. I always think everyone's mixes always sound better to me than my own. A lot of times I think the rough mix sounds better. I break out in cold sweat a lot of times when another major project comes forward. A project they booked me for as one of the mixers, especially now this is no longer a hobby for some time, this is serious business and the client pays a big amount of money for my services. The thought coming up eases me down. A&R managers and artists, and producers don't choose you as one of the mixers if they don't like your sound. It's like they say about magicians, they know how the trick gets executed, the public simply only sees the trick - the magic. On other words, the magician can never see his own trick like the audience can. The audience sees his assistant getting cut in half, but the magician sees the actual trick.

Mixing requires both halves of the brian, left and right or in technical words creative and technical; Law vs freedom, science vs. breaking the rules. So whenever I don't feel creative for whatever reason that day, I start working on the technical side and eventually my creativity kicks back in. No matter what happens I always let the client leave with a good mix.

Any aspiring engineer, audio engineering student, teachers and professors, and engineering colleagues are free to openly discus techniques and views on this and techniques in general by emailing me on royalcollegeofsurgeonsinc @ gmail . com

Enough talk... it's time I get to the point of this blog. I'd like to share one of my mixing sessions with fabulous and very gifted UK singer/ songwriter Keisha White to the public.

SONG: "WRONG AND RIGHT"

ARTIST: KEISHA WHITE

Producer & Arranger: Harmony for H-Money Music

Writer: K. Nicole White

Vocal Producer: Harmony

Recording Engineer: Harmony

Recording Studio: H-Money Productions, Tottenham (UK)

Mix Engineer: The Music Surgeon for Royal College of Surgeons, Inc.

Assistant:

Mix Studio: Jasonair Music Studio, London (UK)

Mastering Engineer:

Mastering Studio: Metropolis Studios, London (UK)

Executive Producers: J. Dias, K. Saint, and J. Ferguson for J&J Artist Management

Executive A&R Manager: K. Saint for J&J Artist Management

Publisher: J&J Artists

Label: J&J Artists

Distribution: iTunes (mp3 only)

Pro Tools Version: 7.3.1 HD running on a Apple Mac Quad Pro G5

BPM: 122.0

Time Signature: 4/4

Key: F Major Accidentals: C/ D

Duration: 3:40 min.

Console: Digidesign D-Command 24

Monitoring: Yamaha NS10's, PMC TB2A's, Genelec 8050 A's, and Genelec 7000 series subwoofer

Waves Bundles: Diamond Bundle TDM and SSL 4000 Bundle TDM

Intro: Mixing "Wrong and Right" for UK R&B sensation Keisha White wasn't a hard task. It's her latest release after a two year period of media silence. I have much respect for her management team for putting out this track all by themselves. The track is a straight forward R&B song. The BPM speed was quite interesting. The song is produced and recorded on a rather fast speed of 122 in terms of R&B music, but it was composed at half the speed (61 BPM). The reason, make it easier and more natural sounding for the remixer of the song. Which in my opinion is genius! The producer and recording engineer of the song Harmony did a great job getting the emotion of the singer, Keisha White right. He prepared the session for mixing above expectations. The song was mixed at new studio facility Jasonair Music Studio in London UK at the end of February of 2008 entirely "within the box" and mastered at Metropolis Studios in London UK. Both studios are located in London's famous West End area.

During the session all my dynamic processing and sound design happened "post-fader" as I don't want any thing to change on the input when adjusting leveling (gain). On another note I inserted the Waves PAZ Analyzer a few times a cross the board to find annoying "ring" sounds so I could easily and time efficiently eliminate them. Audio analyzers are an extremely powerful tool, both during recording and mixing. You can basically see what is going on exactly on that channel in terms of sound, from good to bad and be able to fix it quick!. However you need to know how to read them correctly and don't depend on it only. Please depend on your ears, cause you are an enginear and not an engineye.

I also ran out of the available Pro Tools Voices (HD 3 system) quickly. My assistant engineer flew the vocal tracks around to free up some Voices by letting some vocal tracks share voices with each other when not played simultaneously at the same time. We needed to downgrade the session half way through the mix session too, from 88.2 kHz to 44.1 kHz which is an extreme sacrifice however it's still CD quality, to gain additional voices and more processing power, which was absolutely needed, but we contained 24 Bit rate resolution for mastering.

At the end of the day I invited the producer of the track Harmony down to the studio to come and revision, tweak the overall leveling and panning of the song with me, as the producer holds the emotion of the song and I like to maintain those.

VOCALS: Waves Renaissance EQ, Waves Renaissance Compressor, Waves C4 Multi-Band Compressor, Waves L1+ Maximizer, Waves Renaissance Reverb, Waves MetaFlanger, Waves Doubler, Waves SuperTap 6, Antaras AutoTune, Digidesign Dynamics III.

The song consists of a few instrumental tracks, but a lot of vocal tracks (harmonies, add libs).
The vocals were my main focus when I started mixing. The harmonies existed of Keisha White's own harmonies and some male harmonies to give them more range and power. The male harmonies were provided by one of the singers from the new R&B male group Fundamental'03 (Geffen Records).

I splitted nearly every single track in the session at least once, but in most of the cases twice – the so called parallel processing chain. On the vocal group I used the C4 Multiband Compressor to keep certain areas in control. As the C4 Multiband Compressor doesn't sound much like a compressor, Renaissance Compressor followed to emulate an old classic LA2A before hitting the L1+ Maximizer brick wall limiter to flatten out the vocal group completely, yet leave some dynamics. On the EQ part I added a few dB at 1,300 cycles and 10,000 cycles to open up the vocals a bit, and made a few cuts at 247 cycles and 2,200 cycles for a more transparent sound. All the vocal parts got threaded differently to give them all their unique signature and position in the mix.

For effects on the vocals I used Renaissance Reverb to give a nice background blend to them. MetaFlanger was inserted to make them sound less static and a bit moving around the centre image like more of a chorus effect with the tape option activated to make them sit more solid in the mix. Doubler 4 functioned as a pitch shifter/ stereo widener to make the vocals sound larger than life and the singer like a super star. SuperTap 6 was used three times in the mix, one was used to give the vocals more depth and blend, and a second one was used (offline) to emphasize some key words in the lyrics phrases as quarter note delays (ducked). I ducked the vocal effects to achieve a clear sound, as delays and reverbs tend to mash out the vocals sound when it becomes almost inaudible. The trick is to place the compressor after the delay or reverb. You set-up another stereo Aux Input track and send the dry vocal group track to the Aux Input track, and do another send from the dry vocal group track to the compressor's external side-chain input. It's a great trick but take in account it will only work on dedicated individual effects rather than global effects.
An extra Aux Input track (pre-fader) was created to act as some sort of "exciter" effect. I heavy compressed the incoming (dry) vocal group signal (-30 dBFS), followed by Renaissance EQ (post compressor the EQ to avoid some low end rumble coming through when its filtered off) with the high pass shelf set to around 770 cycles and large boosts at 5,000 cycles, 7,800 cycles and 10,000 cycles. This trick is also known as MoTown EQ (not the actual device!), and its descent and discovery comes from the legendary 1960's MoTown era. SuperTap 6 acted as the vocal "exciter" with extremely short delay times set over the entire stereo spectrum of the mix. Its phasing effect complements the vocals.

DRUMS: Waves SSL G-Master Buss Compressor, Waves SSL E Channel, Waves SSL G-EQ, Waves TransX Multi, Waves TransX Wide, Waves C1 Compressor, Waves Renaissance EQ, Waves Renaissance Reverb, Waves C1-sc Sidechain Compressor/ Gate, Waves Renaissance Bass, Waves MaxxBass, DigiDesign Sound Generator, Digidesign Dynamics III, Digidesign PitchShifter.

Before I started to clean and polish the drums I set up a few Aux Input tracks to act as instrument Group Tracks. On all of the Group Tracks I inserted the SSL G-Master Buss Compressor and TransX Wide plugins to act as overall compressor and transient emphasizer/ gate. I mixed the drums through a main Aux Input track acting as the drums main Group Track with, yet again a SSL G-Master Buss Compressor inserted to blend them together and give them that nice punch of the classic analogue SSL consoles.

All the drum sounds came of the classic Roland TR-808 drum machine. I splitted nearly all of them twice, with the first one EQ'd only – acting as the ORIGINAL fader, the second one heavy compressed (-12 dBFS) and EQ'd – acting as the MUD fader, and a third and final one extreme compressed (-30 dBFS) and a high pass shelf EQ with a lot of high end acting as the ATTACK fader. Blend together they formed the sound of the drums. To emphasize the transients on all of the tracks I inserted TransX Multi at the end of the processing chain to attenuate the low and high end content of the instruments timbre. On its Group Tracks I inserted TransX Wide to act as overall sound shaper and gate. The wonderful thing about the TransX plugins is that they don't only design transients but also add harmonics to the instrumental content and on the TransX Multi you can do it frequency selective. On the original kick I inserted MaxxBass to make the kick audible on the smallest and most common multi-media speakers (TV, Radio) available on the consumer market, as the 808 kick only exist out of a pure sine wave, I created a fourth mono audio track and added ProTools own "signal generator" (offline) running through the entire length of the song (50 cycles sine wave, like an oscillator on the drop frame of old analogue consoles) to reinforce the kick drums low end content. ProTools own Gate/ Expender III acted as gate, and kick ORIGINAL as trigger (pre-fader). I inserted Renaissance Bass on this signal to add additional harmonics and more power to the low end regions. I dipped around 300 cycles to give space for the keyboard sounds and 600 cycles to lose some of the kick drums energy so it wouldn't overpower other instruments in the mix. Boosts were set at 64 cycles, 130 cycles and 3,500 cycles and 4,000 cycles so the ear can easily find the kick. I didn't split the OH instruments.
During the blending of the drums I noticed that the snares/ claps overpowered the drums, so I added C1 Side-Chain Compressor/ Gate with very fast attack and release times on both of the Group Tracks. The snares and claps wound up to 500 cycles to give the kick more ..breathing.. space and the same for the OH group. I put C1 Side-Chain Compressor/ Gate with a large dip at 700 cycles to give the snare and clap tracks more room to 'breath'. I added a nice Renaissance Reverb on the snares and claps to bring them more to the back of the mix, just behind the vocals. An additional Renaissance Reverb was used lightly on the kick Group Track to make the feel of the song more big and mystic, yet keep the kick clear enough to be upfront and hitting your stomach and chest.
I felt that the snares/ claps sounded a bit weak so I created 2 additional mono audio tracks and placed a copy of the snare on one, and a copy of the clap on the other. ProTools own PitchShifter plugin (offline) was used to de-tune the channels information with about 7 cents to give the overall blended snares and claps more body and strength. This trick basically works on every transient sound that sounds a bit to 'thin'.

BASS: Waves S1 Imager, Waves Renaissance EQ, Waves C4 Multiband Compressor, Waves MaxxBass, Waves Renaissance Bass, Sans Amp, Digidesign Dynamics III.

The bass of the track carries a lot of low and low-mid energy and lies directly below the kick. Another problem was the phasing of the bass. I used S1 Imager (offline) to make the bass mono. I needed to find place for both, the kick and bass, within the mix without fighting each other for the same space; therefore the bassline got placed below the kick drum. The bass got splitted once, and I used the original fader to EQ the bass only, with a low pass shelf around 3,000 cycles and some boosts at 72 cycles, 980 cycles and cuts at 64 cycles to give the kick some space, and 734 cycles to give the snares and claps some space. The splitted channel acted as the MUD channel similar to what I did on the drums to contain more control and phatness.

On the Bass Group track I added C4 Multiband Compressor to control its low end so it wouldn't jump in the song. To distinguish the bassline from the kick drum I inserted Sans Amp for some distortion on the same Group Track. I tweaked it until I got the desired effect I was looking for.
On the original EQ'd only fader I inserted MaxxBass to add some harmonics and make the bassline audible on the small multi-media speakers of consumer audio products. On the MUD fader I inserted Renaissance Bass for additional punch, low end content and harmonics.

ProTools own Compressor/ Limiter III acted as ducking device on the Group Track to push the kick through whenever it kicked in during the song to avoid the kick getting eaten by the bass. The ORIGINAL kick drum fader acts as the trigger (pre-fader). I put the bass group on mute when I continued mixing the song. The reason why? Cause the bass contains so much energy it can literally screw up your mix. I blend the bass back in as the final instrument of the song and tweaked it further to make it fit without interfering with the instruments.

KEYBOARDS: Waves Renaissance EQ, Waves Doubler, Waves MetaFlanger, Waves Enigma, Waves MondoMod.

The only tracks left are keyboard based sounds and didn't need much treatment. EQ served to emphasize the instruments timbre and a boost around 14,000 cycles to gain some air and make them sound less digital and more analogue like the old classic analogue synths from the 70's and 80's.

The groups' chain of effects made the keyboard sounds really stand out. I used Metaflanger to emulate an old analogue Ampex 440 tape machine. Enigma acted as a nice flanging effect and MondoMode made the entire instrument group move around as some sort of a chorus effect. Doubler 2 functioned as pitch shifting and widening tool to give more depth and spectral placement of the keyboard sounds.

FX SOUND: Waves MaxxBass, Waves Renaissance Reverb.

The entire song only carried one single effect sound, a closing door. Because it was a great picked and clean sample I did not use any EQ or compression. I wanted to make the sample sound more cinematic with some phatt low end. MaxxBass with loads of low end and harmonics created that sound I was looking for.

Renaissance Reverb inserted directly onto the channel to bring the sample to the background as if the door is closing behind you.

MASTER BUSS: Waves SSL G-Masterbuss Compressor, Waves S1 Shuffler, DuY DAD Tape, Focusrite D2 EQ, Waves L3 MultiMaximizer.

The masterbuss contains the SSL G-Master Buss Compressor to gain additional punch and glue the mix together with gentle and subtle gain reduction. S1 Shuffler was used to give the mix a nice wide 3D radio ready sound.

When widening the stereo image the human brain perceives a loss in mid-high and low end frequencies. I made up for this by adding some dB at 1,088 cycles – the in your face frequency, and a 1 dB bass boost. DuY's DAD Tape functioned as _" analogue tape emulator at 30 IPS, I added a D2 EQ from Focusrite to shelf about 1,5 dB high end back into the mix to make up for the high end loss during the "transfer" to analogue tape.

Lastly and finally I added L3 MultiMaximizer with the output set to -5 dBFS to gain some headroom for the mastering engineer before we went on printing the four common versions; main mix, instrumental, tv version and the acappella version of the mix.

SIDELINE: Additional Tricks.

Lead Vocals:Split Reverb technique is one of my favourite techniques to apply on lead vocals, to give them more depth and more of a stereo effect.
Create two Aux Input tracks and load the same reverb plugin and the same settings on both tracks. Pan them hard left and hard right. Send the lead vocal to both tracks and adjust the decay time of the reverb on one of them. Tweak until you got a desired 'stereo' effect.

On automation: To give your song more of a 'live band' feel and thus more emotion, ride the kick drum group fader during the song a little by changing the level a bit ever 2 to 4 bars. On the chorus put more of the attack fader in (about 1 dB), so it sounds like the drummer is hitting harder during the most emotional part of the song. Put the snares group fader up randomly throughout the song.

On the final chorus try to insert another EQ (1 band) on the vocal group track. Ride the MHF range with a little boost during that section of the song to really make the vocals 'scream'.

Sample Replacement: Sample replacement is usual used as a tool for mixing engineers to replace the performance of a bad drummer. I usually use this trick to 'enhance' sampled drum patterns, therefor I rather call it 'sample enhancement' then replacement.

You can easily enhance drum samples by ProTools own Beat Detective plugin. I won't dig deep into it but for those who are interested drop me a line at my email given before.

Another trick to reinforce TR-808 snares instead of de-tuning the parallel sound source is to enhance it with a short sampled TR-909 snare, which has more body. Blend the TR-909 snare in BUT keep it low so it won't mess to much with the original TR-808 snare sound. You can do the same trick for the kick drum(s). Put a thumb -low end kick drum sample (needs to have information below 40 Hz), that suites the sound of the other kick drum(s), on a new channel and blend it in until it sounds phat.

Using M/S Creatively during Mixdown!

2008-07-17T20:56:00.000-07:00

Yes you read it correctly: "M/S (MID/ SIDE) technology during mixdown in a creative way. You might wonder how? Or even why? I don't blame you. I have to say I'm kind of a shamed that I just found it out like 48 hours ago via a friend of mine named Stylo, he works for Teddy Riley at Night Bird Studios in West Hollywood not far away from my hot spot Studio A from Will Smiths The Boom Boom Room in Burbank (also Cali).

Let me first refresh your mind on what M/S exactly is and what it precisely does.

1. What is M/S technology ?

Most people who are into recording music know about the M/S microphone technique where you have 2 different microphones for mid and side signals. Recording a signal with M/S microphone technique means to create a stereo (L/R) signal by mixing the M and the S signal together in a special way that will create a very mono-compatible stereo signal. To record in M/S you have to feed the M-signal to the Left and Right channel of your mix and the S-Signal will be fed phase-correct to the Left-channel and phase-reversed to the Right-channel. This means you will have to split the S-signal and use a total of 3 channels for only 2 microphones that will be mixed together to a conventional stereo-signal. The bx_digital does all that work for you, just feed it with the separate M- and S-microphone signals…

For additional details about this technique you may have a look into every decent book about recording basics if you want to learn more about this method to create very mono-compatible stereo-signals, e.g. when recording drums, classical music, choirs or accoustic instruments in general.

2. What is M/S good for in MASTERING ?

Well, it may sound simple, but our bx´s have a built-in M/S matrix that will separate EVERY stereo signal into its mono sum and the stereo difference signals. This way you are able to separately control these 2 signals which can be very useful when you work on a stereo mix that has certain "problems" or if you just want to enhance certain elements in the mix.

Ever tried to cut high frequencies of a mix to reduce the "essing" of the lead vocals and AT THE SAME TIME boost high frequencies of your harmony instruments (guitars, keyboards, pianos, etc… most likely recorded or mixed in stereo)??? Well, with the bx1 / bx_digital and their M/S mastering mode you can do exactly THAT!

There is a reason why all of the big mastering places all over the world have been using this technique for decades – without really telling anyone about it…

(source: http://www.brainworx-music.de)

Now you know those aspects and info I'm going to tell you some thing about how to use it during mixdown in a creative way.

Lets say you got your reverb aux up, your delay aux up, and may be some other effects like a harmonizer may be a pitch shifter. Now what you do next is set up another aux input track and load on there BRAIN WORX CONTROL this is the only plugin that will allow you to get crazy and stunning effects with M/S technique.

I love to play around with the frequency knob and put it around 1,088 cycles. The in your face frequency. I play around with the width knob, usually between 200% and 400%. Some times I want to have every thing in phase and some times my effects tail going out of phase makes some cool effects to... I play around and A/B between Mid solo (M Solo) and Side solo (S Solo). It can give you that nice effect that will place your effects and delays in the right spot. One thing you will have to know is that I compress my effects. The reason why is that I get the control of its tail. As you know some times your effects tail fades out to rapidly or it takes to long. This way I can control it myself at all times.

Back to the M/S you know what it does, I gave you a quick introduction of what I usually do on effects now go and figure some stuff out yourselves! LOL! Post your findings as a reply on this post or hit me up on myspace!

Good luck!

MORE EFFECTS!

2008-07-17T20:41:00.001-07:00

More effects!

I thought I'd share another topic about mixing with my readers!

Vocal Effects
To make sure that the vocals are always upfront, in your face, and have their own room and space to breath; I always go and find that frequency that makes the vocal sound like it sounds, its unique frequency- the vocals main fredquency. Once found I go to every single channel of my mix and insert an 1 band EQ or when avialable my board EQ and dial in the frequency of the vocal and make a subtle dip (cut) with that frequency on all the instrument channels to ensure that the vocals will be there period. Another thing that also always works is adding round 10,000 cycles to gain some air for the vocal. Watch out for around 2,000 cycles that usually is an anwanted harsh sound. I'd prefer to sidechain -dynamic EQ around that frequency instead of doing static eq. Also around 240 cycles can give the vocal unwanted sound, not always, some times around that frequency enriches the vocal sound and makes it warm. Listen and decide for yourself. Make sure to make room around 240 cycles for the vocal in that case and dip a little with that frequency on the instrument tracks of your mix.

A lot of times you use effects on vocals which over time fade out to rapidly cause of the dynamics of your dry vocal signal. I always compress my vocal effects to sustain my effects "tail". A side from compression, I also sidechain my effects with the use of my dry vocal signal. The reason for this is that when making use of effects on a rapid instrument, like vocals are, it makes a mess of delays and feedbacks as heard on reverbs and delays (especially the last one). Because of that the vocal often becomes inaudible, We can solve this by using the sidechain "ducking" technique, with the compressor set post effects.

It works quite simpel; Create anorther stereo Aux Input track. Send the wet vocals to the input of the stereo Aux Input track and make another send from the dry vocals to the compressors sidechain. That's it.

To make my vocals sound wide I use Waves Doubler 4 on the back ground vocals and Waves Doubler 2 on the lead vocals (shorter spaced and less detunes). On top of that I make use of delays and reverbs. My delays on the lead vocal are, on the left eight note delays, and on the right quarter note delays, using the standard grey colored plugins from digidesign. Quarter notes sounds better in chorusses, and eight notes on the verses as these are song more rapidly. Both make a great "bouncing off the wall" effect.

For reverbs I like to use a blend of Waves Rverb and digidesign's own grey colored Dverb (Reverb vs. Tail). What I usually do is, I set up two stereo Aux Input tracks. Both loaded with Waves Rverb and both with the exact same settings. One panned left and the other panned right. On the left one I change the decay of the reverb (less or more depending on the sound I want). This effect creates a nice stereo sounding reverb on the background of the vocals.

Tip: Male vocals will sound better with the use of a bright sounding reverb because of the character and color of the males voice, which is dark (low) in general. The opposite applies for the female voice.

I do EQ my vocal effects and any other effects at all times. My EQ is always set post-compressor and all my dynamic processing is set post-fader as most of my readers have noticed and will know. I do this to carve out unwanted frequencies and polish my effects more to suite my instrument, or for spectral reasons. I do filter off lows and highs, especially with digital delays and reverbs as they tend to sound ugly beyond about 7,000 cycles. Another thing I always do is notch out around 1,000 cycles for the same reason and because this is also your in face frequency area which I'd rather have clean to make my music sound more natural and understandable.

Calculating Compressor Release Times
To me this subject has never been discussed neither, addressed properly. Face it, when it comes to compression we're pretty much old school and stone age-ish.

Here's a little trick that I use all the time on my master bus. It'll definitely benefit you and your mixes. Let's say we work in a tempo of 120 BPM, and a time signature of 4/4th. We are working in a gird and bars and beats environment. Now we all know how to calculate delay times or got tought this trick. If not; You can easily calculate a range of interesting delay times if you know the tempo of the music. And of course, the way music is made these days, the tempo is on the display right in front of you (And did you set that by feel? - whole other story there!) So take that number and divide it into 60,000. 60,000 is a 'magic number' derived from 60 seconds in a minute and 1000 milliseconds in a second. Say your tempo is 100 BPM (beats per minute). 60,000 divided by 100 is 600, so we have 600 milliseconds. This is the base figure that will give a delay that is almost certainly too long. But try dividing it by 2, or by 4. Try settings of 300 ms or 150 ms as your delay time. You will find that the delays fall exactly in time with the beats of your music. Want more...? Try dividing by 3, or divide by 3 then multiply by 2. So you have 200 ms and 400 ms. You now have a delay with a triplet feel. You can try one-fifth, two-fifths or three fifths (even four fifths if you like), giving 120 ms, 240 ms etc. More complex, but still related to the beat of the music.

OK with this information we can apply the very same formula to calculate compressor release times. We want to get the release of the compressor to recover just on time, just before the attack of the next kick drum kicks in again. Our tempo in this example is 120 BPM. So 60.000 / 120 = 500 ms. 500 ms is the release time of the compressor. This trick some times is also suitable for individual instruments but keep in mind to rely on your ears at all times, after all you are an enginEAR and not an enginEYE!

This is it for now. Please reply and discus this topic with me. For questions please reply to this blog post as well.

Yours truly signing off from an cold UK.

ARDOUR THE FUTURE... MAY BE?

2008-07-17T20:39:00.001-07:00

Ardour... many of you won't know a thing about this great open source sequencer made by Paul Davis from Solid State Logic and Linux Audio. Recently one of the world largest engineering pro schools sponsored the Ardour foundation and started to teach Ardour at their colleges. It's now part of their curriculum.

It cost me quite a while to finally dig into this free open source sequencer from Linux and above all accept its existence, two years to be exact. I first got introduced to it by Prince's engineer when I was working along side him on some George Clinton material and Rosie Gaines (New Power Generation) never finished album. He installed it on an old PC running both Windows 95 and Linux. He insisted that I took at look at it and played around with it. He is a great fan of it and to be quite honest I'm starting to become one too. I'm against monopoly companies that can do whatever they want and I finger point to Digidesign's Pro Tools for a moment. (Yes I know its morally and politically wrong but for once just lets face it.) Not to mention their HD cards that are way over the top when it comes to its retail price. Using Motorola 5600 chips which cost literally a penny and selling them for way more to the end consumer. Unfortunately a lot of other pro audio manufacture companies are following this but Pro Tools stays the industry standard. I support Ardour because its open source and we all can change and add features which we want to be there, no restrictions! It makes the sequencer so much more powerful and flexible then any other D.A.W. sequencer on the surface of the earth, and yet better you don't need to pay for! Because it runs on Linux its more stable too. It's GUI looks familiar to those using Pro Tools, Nuendo, Logic, Cubase etc. which is an absolute pro. A con however is the processing, it is low latency but I need a lot of processing power when I'm mixing. I'm currently looking into buying Apogee's Symphony, a total of 3 cards which will allow me up to 192 channels of DSP power and 96 compared to 96 channels of DSP power in a HD 3 system on 192 kHz and it cost a fraction of a HD 3 system and way better quality. You can run 200 plugins compared to 96 on a HD 3 system. If this is not possible I'm going to have a look in building my own DSP chip set cards for Ardour using Shark chips which are way better and way more efficient then the traditional Motorola 5600 chips. This way I can bypass Pro Tools and run a more efficient D.A.W. sequencer and still have the flexibility of routing my channels to different programs and systems. THATS WHAT I WANT! Not to mention your system can run native plugins on external DSP power which means its CPU efficient and on top of that you save about 50% on all your plugins.... so I would load them up with my Waves collection of plugins. After-al companies pay a license fee for using the TDM source code. Now you can work CPU efficient, have more control and flexibility, and on top of that its cost-efficient but still be able to walk in to any major recording facility and connect your system to their system and distribute your Ardour tracks to Pro Tools and Logic and so on. You do the math. I'm sold with Ardour connected to my Mac Pro running Pro Tools HD3 but I think I even can get away now with Digidesigns entry level HD1 and have much better quality of audio processing!

Lets list it:

Free Professional Audio Sequencer

Open Source

Linux stable OS

Better quality DSP processing

Better A/D D/A Convertors

Better Sync/ Clocking

Cost Efficient

NOTE: It uses compiled coding so some knowledge about programming in C++ is required, but to actually install and run Ardour with Jack Audio and JAMin they wrote the command lines for you in a step by step instruction manual.

http://ardour.org/

THE IGNORANT....

2008-07-17T20:38:00.001-07:00

What really bothers me the past few days is the endless talk of the ignorant especially those of A&R managers. The choice for these select group of people when it comes to mix engineers is some what hilarious. I've dealt with A&R managers from almost every corner of this globe. It is good to have them on your side, don't get me wrong on that part. They can work wonders when it comes to mixing gigs. But time and time again I have to hear how they select who will get the mix job. They go to Billboard.com and check which record is currently topping the charts. Then they go and find out who did it. Usually this is the same selection of mixers which I will never speak bad about cause they really do a fantastic job and on top of that one of them has become a quite good and close friend of me and his manager and I are close friends. He even provides me with gigs every now and then. But to get back to the point, the A&R manager and the other group of ignorant people choose a mixer based on what record is currently topping the chart. They fail to realize every single time that a hit record already is a hit record when the producer has done his job. They still do believe that the mixing engineer can make or brake records. In a way this is true, but that is only technical but for 90% the record is already made by the producer and is hit record material or not. I can't change what is already there, but they believe different. This is why I 300% and 1,000,000 times more rely on the producer. He gets to decide either before or after the A&R manager who s/he wants for the project. To the rest of the world I always explain what we actually do as mixing engineers. But first I need you to understand that we CANNOT make a hit record. The producer is the one that needs to have produced already a hit or else it won't. So now you know we can't make a hit record. Why would you possibly invest in a mix then? Good thinking. Picture this. You have a brand new top of the line Bentley, great car and a real hit. But the car is dirty, it is covered in mud cause you've been driving around in it. Pretty much a shame cause its not getting its full allure. You bring it to a car wash and get it cleaned so it shines again. Now this is what a mixing engineer does and think of the mastering engineer as 'just' the guy/ girl that fixes a broken tail light.

Hope this blog clears all the clouds and won't leave behind an empty cloud but instead encourages you to reconsider and think about what we do. Besides I once remember that a famed producer in the name of Ron Fair said that music is unexpected and to stay fresh it needs to have a new and unique input.... on top of that friend and master mixer Dave Pensado said that sounding fresh is better then good, but A&R managers are no risk takers, unfortunately, so it's up to the producers out there to put together a great team of creative people.

ANTI PIRACY AND ITS COPY PROTECTION

2008-07-17T20:36:00.001-07:00

I thought this would be worth a discussion; Anti piracy.

The music industry keeps on yelling that mp3's are decreasing record sales and they keep on investing serious amounts of financial resources in encrypting mp3 data. As we all know this has failed every single time. As software programmers and engineers can tell you for every software programmed copy protection attempt there's a hacker able to brake the encrypt code. With that knowledge I still don't get why the music industry keep on trying over and over again when we all know it will fail.

But remember for every downside there is a positive side. The mp3 era has also given acts, artists and bands the opportunity to present themselves and even some got their major breakthrough. Remember Usher's recent number one hit song "Love in the Club" and the story behind it? This is a good example on how mp3 can work as an advantage. The producer leaked the track on the internet cause the record company wasn't feeling the track. Within weeks it became a huge success and eventually when picked and finally released it hit the number one spot on the Billboard Hot 100 chart.

Interesting enough this allows a discussion to take place which can't prove right from wrong and can't be wrong and right; They lost record sales due to the leakage. But then again without the leakage the track probably never saw the light of day. The record companies will put their focus on the first statement.

iTunes is the best selling (online) record store in the world with billions of sales. Billboard dedicated an entire chart to mp3s and even ringtones. It seems like the recording industry can't face the facts and can't accept them so that they can move on and spent their money right.

Ever now and then when I'm working or when there's a business related gathering the copy right protection/ anti piracy issues comes along and to be honest the creative people do have the answer.

Why not spent all the financial resources they are spending right now on encrypting mp3 data, to contract a manufacture in the field of audio to come up with a brand new medium, not a format, a medium. We all know the revolutions from vinyl to tape, compact disc to mp3. And don't we all remember every single time a new medium came along the industry stood up and yelled a negative thought?

Lets say the industry is never going to agree on contracting a manufacture to develop a brand new medium. Lets face the facts and make the music free.Yes you read it right, FREE. Lets focus on every thing surrounding the music and change our selling strategics. The world has become visual dependent. We all love music videos. We don't like albums any more, we like singles.

OK facts: Music Videos, Singles. Lets create an entire album consisting out of single material only. Every single track on the CD should be suitable as a single release. Be strong on its own. Album is done. But wait thats only the music and thats going to be free? Correct. Now lets focus on the visuals. If every single track is suitable for single release; lets make music videos for all those tracks and compile it into a DVD. Visuals are not free only the music so the DVD is for sale. So the deal will be, buy the DVD and get the album for free. And every one loves freebee's.

This is just one example of what could be a great solution.

Lets try and get the recording industry to put their money were it eventually will become a profit.

Signing out,

The Music Surgeon for Royal College of Surgeons, Inc.

The Way We Split Instruments

2008-07-17T20:35:00.000-07:00

THE MAGIC OF MIXING MUSIC

As a response to my earlier post I'd like to share some of my most common used chains that establish my sound.

I'm a big fan of a software plugin processing developer named Waves. However, don't get me wrong; I'm also a big fan of analogue outboard gear and especially flagship consoles. You simply can't get the sound of a large flagship console like those from Solid State Logic. You simply don't have all the routing circuitry and bussing. The emulated software plugins only provide the circuitry of a single channel strip. This is one of the reasons why I like to route my Pro Tools channels out to the SSL 9000J SuperAnalogue series console. I do a lot of submixing in Pro Tools to get a more organized mix session layered out on the desk. If in case I got additional outputs I will route my Pro Tools SFX Aux Input tracks out to the small faders of the desk for the same reasons. All the group channels I like to threat with a blend of outboard gear as there is still a notable difference in sound between digital and analogue processing. One big advantage with the digital world is that we are able to work on several projects simultaneously as every thing gets recalled for you within a fraction of a second. Meanwhile in the analogue world you either need to watch a little screen on the console and re-set the values which, especially on a large desk can easy eat up ten minutes of your time. Not to mention all the outboard gear and patchbay needs to be re-set manually by using recall sheets and every single time your mix sound will change without a doubt. Only if you're lucky and own your own room you can leave every thing overnight. Look at the digital world as leaving your mix session overnight.

What I like to do is switch between passive (opto) and active (electric) processing and solid state and tube processing as well as tape saturation. In more understandable words; divide my frequency spectrum in areas to threat differently by the use of electric circuitry. To keep it simple think of this as a multi-band processor. I'll save you the story about harmonics in this post to keep it easier to read and understand.

LF: On my low end spectrum I like to use API's 550A EQ's for coloration as I like its pre-fixed settings -off boosting, SSL 292 EQ's (G EQ) for filtering and coloration as its resistors pre-boost dip along with a slight pre-cut rise to give my low end a distinct sound. The Waves Renaissance EQ's and Compressor's for additional coloration and filtering and on top of that the IC input and twin VCA based SSL G -Master Buss Compressor as found on the classic 4000 series console. I often find myself also attracted to using API's 2500 compressor using traditional feed back circuitry, hitting it extremely hard. The punch resulting is some thing I find very desirable on both the SSL and API compressors.

Other then just mixing I like to sound design a lot as well. I recommend Waves TransX above SPL's Transient Designer 4. This is because of control and harmonic issues which I wont dig into. On the splits I use TransX Multi to threat different frequencies and on the group I use TransX Wide to control the overall sound of the audio program and TransX Wide acts kinda of a gate too. Tape compression will help getting my low end to sound phat and solid with a post high frequency rise shelf EQ of about 1.5 dB to make up for the loss in high end content caused by the tape compression. My low end area sounds best when processed passive.

So I have given you an example of how my low-end sounds. I now want you to try it out for yourself for the other frequency areas as listed below and find out what sounds most pleasant to you. Post your findings via reacting on this blog post or by email: royalcollegeofsurgeonsinc@gmail.com

MF: On the mid area solid state designed products really come to their fullest extent in my opinion. My mid frequency sounds best when processed passive.

MHF: This really is a combo of solid state and tube processing. My mid high frequency sounds best when processed active.

HF: High frequency response best both vintage and high-tech especially products from AMS Neve, Pultect and Tube Tech will really sound at their best in this area among the normal plugins from for example Waves. My high end area sounds best when processed active.