Saturday, August 31, 2013

Trent Reznor releases two versions of new album "Hesitation Marks"

On September 3rd, Nine Inch Nails will release it's first effort since 2008 entitled "Hesitation Marks".  This new album marks the first major label release by the group since 2007's "Year Zero".  If you don't know who or what this band is, here's a link to get you more accustomed with this band and what they're really all about (shame, shame on you).  Any way, Trent Reznor has been at the forefront of the music industry since he first garnered mainstream attention back in the mid 90's.  Back then it was his use of samples and his use of the "new" digital technology that made others refer to him as a  genius of the new movement of music.  He has always been at the forefront of technology and how it can be used to further the art form.  It wasn't until recently that he was able to innovate in other areas of the music industry, most notably, distribution.  In 2005 he released his final album for Interscope, "With Teeth".  After he finally fulfilled his contract with Interscope, he decided to explore the different avenues of distribution that were available at the time.  The internet was taking over and Reznor knew it.  He recorded his next album, "Year Zero", entirely on his Macbook using Native Instruments, Reason, and ProTools exclusively.  Reznor then released this album to the public mainly via the internet.  He decided to explore the medium by actually intertwining the distribution with the concept of his new album, which was set in a future, dystopian world.  After seeing one of his contemporary's, Radiohead, distribute their newest album through their website through a pay as you go model, he decided to one up them and give his next two albums away for free through his website.  These two albums, although freely distributed, ended up being fairly critically and commercially successful.  He launched a massive tour in support of both albums, and eventually retired in late 2009.  We all know what he's been up to since then...

Oscar winnin' Mother fucker
 So, amazingly enough, he decided to come back and make music with Nine Inch Nails and announced it earlier this year.  To fanboys like me this was the greatest news ever. 

Nine Inch Nails waved Good bye already.  

So, of course with his newest album, he decided to shake things up again.  Was he going to give it away for free, hell no that's been done.  Would he let you name a price... Go fuck yourself.

This time he decided to take on the loudness wars!!!!  How was he going to do this you ask?  Well, of course he'd release two separate versions, one "loud" version for the sheep who use iTunes and buy CD's, and one "audiophile" version, for us dorks.

Get in my van and listen to my audiophile

 The audiophile version (not kidding, it's actually called that) was mastered in a different, more dynamic way than current music is. The loud version is apparently mastered just like that, loud.  It should compete with everything everyone always hears, and should reproduce itself nicely on most systems and speakers that people listen to on nowadays (I'm fucking looking at your Dr. Dre).  The "audiophile" version will be noticeably different on higher end systems, and is offered for free with purchase of any other version of the new album, "Hesitation Marks".  You can get more info on it here, and you could still listen to the album for free at iTunes until it's released on September 3rd.  When that day comes, it's all up to you I guess.

Friday, August 23, 2013

Creating Tempo Synced Delay Manually

One of the most commonly used effects in music is delay.  Delay is a simple time based effect that can be utilized to create a plethora of different sounds and textures.  In the old analog days of recording, before the advent of digital delay units, delaying the sound was done by hand.  In order to create a feedback delay, they did just that, by feeding back the delayed signal to the input in order to be recorded onto tape once again.  With the advent of digital recording and all the fancy delay plug ins that have come with it, it is quite easy for us to apply, and sometimes overuse, our delay effects.  One very popular delay technique is called tempo syncing.  This is a delayed sound that stays in time with the music, typically on 1/8 and 1/16 notes between beats.  In the digital realm this is quite easy since almost every plug in comes with a tempo sync option.  You simply click the sync button and set the amount of delay you want accordingly, but there is a catch to this.  There's no free lunch, and there's not always automatically tempo synced delay, especially for those of us who mainly record live instrumentation.  The reason is that not everyone records with a click track all the time.  The plug in needs to connect with you DAW's set click tempo in order to be able to sync up it's delay with different subdivisions of beats within your track.  Luckily for us, there's MATHS!

Now, if you don't know the tempo, or bpm (beats per minute) of your song, don't fret.  Listen to your track carefully and keep your eye on the clock, you're going to figure this one out with your brain, I know mine could use the exercise.  Count out for ten seconds, you won't be exact, but you'll be close enough to fine tune your calculations later.  Say you counted out 10 solid beats within your ten second time frame, here's a calculation to figure out your beats per minute.

10 * 6 = 60bpm (10 beats per 10 seconds equates to 60 beats per 60 seconds)  Simple enough.
Now, we need to calculate the length of time per beat, then from there we can subdivide that accordingly.

60sec / 60 bpm = 1 second per beat of music.

Awesome.  Since most of our delays use milliseconds to set delay time, we need to know how many milliseconds are in a single beat of our music.  This is easy once you know how many seconds per beat, which we just did above. 

1 second * 1000 = 1000 milliseconds per every beat.  So...
60 / bpm * 1000 = ms/beat.  Yea!

So if you set your delay time to 1000 ms, you will have a delay of exactly a quarter note.  500 ms gives you an 1/8 note delay, and 250ms gives you a 1/16 and so on.

Here it is in action.

This is a delay of 1000 ms.  Notice how this doesn't work for this track since each note is played on the downbeat.  The delayed notes mix with the played ones, sometimes creating an interesting harmony, but mostly sounding dissonant.



Here's a 500 ms delay with zero feedback.  This would be an echo an 1/8 note after the beat.

The effect above is kind of boring, I must admit.  So if you want an interesting sounding echo/delay, I suggest using dotted subdivisions of notes.  That means a beat and a half of that beat, so a dotted 1/8 note would last an 1/8 of the beat plus another 1/16 after that, since 1/16 is half of 1/8.  Add some feedback and depth and you get a pretty interesting sound.

Monday, August 19, 2013

Parallel Compression explained (Part 2)

In my last post I gave you guys a run down on the function of downward compression, now let's get into the real fun and complicated stuff.  Upward compression.  Phonetically, linguistically, and logically the words make sense together, upward as in compressed from the bottom instead of the top.  Imagine the bottom as the bottom of our input-output plot, and the top as, well, the top.  The bottom of the graph correlates to the softer parts of the signal, while the top is the louder parts.  So upward compression is basically taking all sounds UNDER the threshold and BOOSTING them by an amount determined by ratio.

Upward compression plot showing a threshold of -17.5dB.  Notice how the louder sounds are unaffected while the softer ones are boosted.
Now, the question is, what makes this any better than regular old downward compression?  I mean, they're essentially accomplishing the same thing with the dynamic range of the signal, right?  Yes, they are, but louder portions of signals contain very important transient and sound envelope information that are ultimately affected by downward compression.  Thus, upward compression gives us the best of both worlds; unaffected transients and a more malleable dynamic range that we can play with.  Unfortunately, machines that can do this effectively are few and far between and that's without mentioning the fact that with this operation your noise floor rises from the grave, rearing it's horrific head.  For example, think about a noise floor of -80dbFS and a threshold set at -20dBFS.  Now even with a gentle 2:1 ratio your noise floor will be raised to a very intrusive -50dBFS.  So now you understand why the practical application of such a device is unrealistic.

This is where parallel compression comes into play.  By mixing a dry, uncompressed signal with a compressed signal you can mimic the activity of upward compression, even though it isn't true upward compression.

You'll get a plot that looks more like this with one parallel compression track

Now, for those of you who want, or care to have, a real expert explanation of this go here.  Otherwise keep experimenting with it on your own and be sure to check out my video tutorial on how to set it up in Pro Tools.  Happy mixing!

Saturday, August 17, 2013

Parallel Compression explained (Part 1)

So, in my last post I gave you guys a sure fire way on how to set up parallel compression on any track you like in Pro Tools.  The how to do it is basic enough, send one stereo bus to a new auxillary track and throw a compressor on it and begin tweaking knobs, but the why is a bit more complicated.  In order to fully understand exactly why we use parallel compression, or NY compression as it is commonly called, you need to know that there are two major types of compression used in audio, upward compression and downward compression.

Downward compression is the most ubiquitous kind of compression out there.  All audio and mastering engineers know this kind of compression inside and out.  The main characteristic of downward compression is the function of making the louder portions of a signal quieter.  This is usually done by setting a level known as the threshold, but on some machines and plug ins this value is determined by a mix of a set of input and output values, for now I'll focus on the threshold control since it's most common.  The threshold is a level that is set by the user that tells the compressor at what amplitude, or volume level, the compressor should begin to compress, or become engaged.  Any signal that goes above this threshold amount is compressed at a ratio usually set by the user.  This ratio determines how much of that input signals level is attenuated above the threshold amount.  A 1:1 ratio produces a linear output, thus what goes in is what comes out.  A 2:1 ratio will compress, or attenuate, the portion of the input signal that is ABOVE the threshold by a ratio of 2:1, but only the amount of dB ABOVE the threshold will be attenuated.  Therefore, if an input signal goes above the threshold by 2dB and the ratio is set at 2:1, the output of that signal above the threshold will be only 1dB, essentially halved.  A 4:1 ratio would output the signal at .5 dB and so on.
An input/output plot showing the action of a compressor with a threshold set at -20dB.  Each color above the threshold represents the output of a different ratio.

 This seems simple enough right, a downward compressor is shrinking the dynamic range of a signal, or instrument, by a set ratio by making the louder sounds quieter and the softer sounds louder relative to the overall output of the signal.  Now, you might think to yourself, "but I thought compressors made things louder."  Very simply they don't.  Any form of downward compression is not really making anything louder, it is only squashing the dynamic range.  But, since the louder parts of the signal are attenuated, you can use make-up gain on the output of the compressor to either restore, and even boost, the loudest parts of the signal.  Therefore, compressors don't themselves make anything louder, but by shrinking the dynamic range, they can give you the illusion of things sounding louder, sometimes even punchier.  This is sometimes, but definitely not always, a very useful effect.

Graph of the same signal from above, but with a 7.5dB make up gain set.

In my next post I'll explain the functions of upward compression and how these two kinds of dynamic processing relate to the usefulness of parallel compression.  Happy mixing.

Thursday, August 15, 2013

Setting up Parallel Compression in Pro Tools

Parallel compression, aka New York style compression, is a style of dynamic processing that uses a mix of two stereo busses, one dry and one processed.  It's a popular technique that's used in many professional studios because it gives the mix engineer the ability to bring up the levels of the softer signals without sacrificing the all important loud, transients.  This is a brief tutorial explaining how to set up and use this technique in Pro Tools, but the concepts can crossover to any particular DAW you may use.  My next post will go more in depth as to why parallel compression is used and what makes it different from just regular old compression.  Enjoy.


Saturday, August 10, 2013

Using delay to widen the stereo image

Sometimes during recording we'll make the decision to record certain instruments with a single microphone, most likely if we're trying to get a more natural, dryer sound from the source.  This is all fine and well during tracking, but when we get to the mixing phase we find ourselves wanting more from that instrument.  Having only one mono track to represent an instrument doesn't give you a whole lot of options when it comes to placing it in the stereo field in regards to it's depth and width within the mix.  In this tutorial I'll demonstrate how to use the Haas effect to give you more options when mixing your mono tracks.  Enjoy.


Using playlists and loop record to edit tracks in Pro Tools

Ah, fixed the watermark problem by updating my software so this tutorial should be a lot easier to watch.  I wanted to show you guys how to easily comp tracks in Pro Tools using two simple functions in conjunction with each other, loop record and playlists.  This is a technique I've used a lot and it's such a huge timesaver when overdubbing.  I highly suggest using this trick the next time you need to overdub anything.  Enjoy.


Saturday, August 3, 2013

Fixing Clipping in Pro Tools

Ok, I decided to try a video tutorial for my next post and this is my first time ever trying this so we'll see how it goes.  In the video I show you how to fix the nasty sound of digital clipping inside Pro Tools.  Sometimes it happens and due to deadlines or a slew of other reasons we have no choice but to try to fix it in Pro Tools.  Here's a technique for fixing clipping and peaks that I use a lot that I think you guys could use as well.  Oh yeah, this is the demo mode of the software and there's a watermark throughout the video that will remind you of that, sorry.