My Personal Top 20 Songs (Part 3)

Continuing, still not in any order other that what I feel like writing about today…

11. Skinny Puppy – Smothered Hope – Source: Remission EP (1984)
Skinny Puppy was my introduction to “alternative”, and the whole concept that there was music being made that you would never hear on the radio in a million years (even though it had mass distribution, lots of sales, and a dedicated fan base), and in many cases, it was truly great. It didn’t hurt that they were from Vancouver. It also didn’t hurt that between me and one of my best friends, we had almost all of the gear that they used to create their earliest work. Obviously, we covered their songs, and this one in particular sounded impressively close to the original. I’ve never been able to put my finger on what exactly it is about this song I love so much, but it never gets old for me. I even quoted from it in my high school yearbook. FYI: Yes, that drum machine is a Roland TR-909, and yes, this was many years before everyone, even Madonna, started using it. I think that was the “jump the shark” moment where my friend decided to sell his.

12. Kings – This Beat Goes On/Switchin’ To Glide – Source: The Kings Are Here (1980)
Yay! Another Vancouver band. I swear these are all coincidences, as I never lived in Vancouver until 1999. Believe it or not, I found this album (on vinyl) in near-mint condition in a recycle bin in New Westminster, around 2003-ish. This song is the textbook for rock keyboard players. It’s especially nice that you can pan the whole track hard-right and get a pretty good isolated listen to just the keyboard parts. It has the holy trinity of rock keyboards: schmaltzy piano, hammond rock organ, and mini-moog synth lead. Of course I know how to play this song, but I’ve never been in a band willing to cover it. I love getting two songs for the price of one, and I love how they get away with rhyming “wanna” and “Toronto”.

13. Stealers Wheel – Stuck In the Middle With You – Source: Stealers Wheel (1972)
I’ve loved this song since my earliest childhood memories. Listening to it now, I still find it hard to believe that it was recorded in 1972, as the sound quality of it outshines just about everything else from that era. Maybe that’s because it came out of Apple Studio (not Abbey Road studio…the other one, where Let It Be was recorded), and George Martin was probably involved. I’ve been a sucker for hand claps my whole life, and this song is probably why. But it’s the bassline that makes this song for me. I spent a crazy amount of time around 1994 painstakingly programming the whole bassline into a Boss DR-550 drum machine, and I can tell you, it’s WAY more intricate than you would believe, just casually listening to the song. Around that time, I had the song stuck (hah!) in my head for several months. I would wake up in the morning with it playing in my head every day, and I thought I had gone insane and it would never leave. I’m always just a little nervous about that whenever I hear it again.

14. The Cars – Drive – Source: Heartbeat City (1984)
I’ve said it before, and I’ll say it again: Love or hate Mutt Lange, as a music producer, you either listen to/read every word he says, or ignore it at your own peril. This song is a beautiful example of why. This is a rare case where the lyrics are a big part of the emotional connection I have with a song. Oddly, I can place myself in the shoes of both the singer, and the person being addressed (and not just because I haven’t owned a car, nor had a drivers license for most of my life). This is another intricate song that I’ve deconstructed and painstakingly covered, right down to the stereo panning on the tambourine. A large part of the original was created using a Fairlight CMI, (a state of the art Series IIx, IIRC) one of the world’s first music computers, which I desperately dreamed of owning for many years. The “Adoption” section of that Wikipedia article linked above reads like a laundry list of my favourite artists, and yup, even Trevor Horn (see #3 in my top 20) shows up.

15. Gordon Lightfoot – Wreck of the Edmund Fitzgerald – Source: Summertime Dream (1976)
See also this great video. According to this interview, Gordon Lightfoot himself considers this the best song he’s ever written. I love this song so much, I can hardly describe it. Even if I listen to it 5 times in a row, the hair on my arms all stands up every single time. The lyrics, and the way it faithfully recounts an actual event while still being entirely musical is a feat. It’s also genre-defying. You could say it’s a folk song, but the rock guitar in it gives it more weight. The arpeggiated synth part around 2:30 (2:45 of the linked video), right up to the climax of the “story” is so trivial, but so critical to the success of the emotional impact of the song, I consider that pure artistry. Whenever I pause to consider what “Canadian culture” is, this song immediately comes to mind.

Only 5 left!

My Personal Top 20 Songs (Part 2)

Continuing with the list. Keep in mind that these are in semi-random order…

6. M – Pop Muzik (12″ Mix) – source: 12″ single (1979)
A friend of mind had this single when we were around 9 or 10, and the whole concept of it blew my mind. Here was a full-sized 12″ record, but you played it at 45 RPM, and there was only one song. The idea of an extended remix was also new to me, since Duran Duran wasn’t a thing yet. Two things in my life probably came from this particular song. First, becoming a DJ a few short years later, collecting my own 12″ singles and playing them in front of hundreds of people. Second, deciding that saxophone was my instrument of choice in school band. There are so many odd connections in my life to this particular song that I could write a book. The short version: This song is at the start of side one of the first blank cassette I bought, the first song I burned to CD-ROM, and the first track transferred to blank MiniDisc. The reason I bought a MiniDisc recorder was to bootleg U2 shows when I knew I was finally going to see them, after totally missing the ZooTV tour. I saw (and booted) 4 shows on the PopMart tour, in Winnipeg, Edmonton, Minneapolis, and Montreal. Guess what the opening “hype music” for that whole tour was, as U2 came on stage? U2 – PopMart Intro (Mexico City) What an awesome song too, by the way. I never, ever get tired of it.

7. Mark Snow – The X-Files Theme – source: too many to list (1993, 1996 extended version)
If you’ve seen my CD shelf, you might notice that down near the end, there a whole section for ‘X’ that has tons of versions of X-Files theme remixes, and X-Files music. To my knowledge, I have all the X-Files music that’s ever been released, even the rare, numbered multi-CD box sets with hours and hours of score queues. Music in movies, tv shows, and video games has always been something I pay extremely close attention to, and everything Mark Snow creates raises the bar for what “background music” could be. I’ve done a number of covers of the X-Files Theme itself, but generally, I often find myself thinking “how would Mark Snow approach this?” when I’m working on particular types of composition. Asides: Whenever I get my hands on a keyboard with a good piano sound, one of the first things I do is set up a stereo delay on it, so I can play the theme on it. Also, I love songs that have a strong 12/16 meter superimposed on top of a standard 4/4, which this song is a prime example of.

8. Nazareth – This Flight Tonight – source: Loud ‘n’ Proud (1973)
This is one of the first songs I loved as a little kid, and if you’ve read my Intro post on this blog, you know that story. It was my introduction, specifically, to what I call the “galloping bass”, which is simply playing on the 1st, 3rd and 4th 16th notes in every quarter measure. That galloping bass kills me every time, and I love most songs I’ve ever heard that use it. I also love the total meter change at around 2:15 (the “doo-wop” section), the haunting guitar wailing, and the relentless hi-hats.

9. New Order – Bizarre Love Triangle (Shep Pettibone Remix) – source: 12″ single (1986)/ Substance 1987
Surprise! This song has that “galloping bass” too. I loved playing this song as a DJ, and it was what I call “intelligent dance music”. I’ve covered this song, and what a project! This song taught me a lot about drum machine programming, synth sound crafting, sequencing, and especially the importance of stereo placement. Every sound in this song moves. And those sounds! This song has the best use of a vocoder I’ve ever heard, and I almost convinced myself to build one (I had schematics for it) just because of it. An interesting thing I discovered about this song is that the chords in the chorus, while seemingly simple, are definitely not. There are two sounds playing entirely different chords that should clash horribly, but don’t. When I was figuring out that part of the song, and just playing those chords together by themselves, I kept thinking “that can’t possibly be right, can it?”…but it is. Last, but not least, my singing voice, as I’ve been told, and honestly believe, is extremely similar to Bernard Sumner, so I can sing this and other New Order songs very well.

10. Peter Frampton – Do You Feel Like We Do – source: Frampton Comes Alive! (1976)
Once of the greatest concert performances/recordings of all time. I have a surround sound mix of this on DVD-A that is even more incredible. Of course, the talking guitar is the biggest highlight of the song. I also love the decision to fully integrate the sound of the crowd into the recording, making it so immersive. The keyboard solo, by Bob Mayo, from around 4:10 to 5:35 is, well, inspiring and depressing at the same time. After years of trying, I know that I’m just physically incapable of playing that well. Still, it gives me something to strive for.

My Personal Top 20 Songs (Part 1)

This is my current personal list of top 20 songs. These are the songs that have had the most impact on me, shaping my taste in music, and in some ways my whole personality. This list hasn’t needed to change much in the last decade or so, since these songs have had incredible staying power with me. Of course, I have a top 100 list as well, that is even more diverse, and includes an insane range of musical genres, and spans centuries. But these are the ones I go back to time and again, because of both the quality of the songs themselves, and the personal connections I have with them. I’ll be describing a bit of this in the next few posts. So, in no particular order:

1. Afro Celt Sound System & Peter Gabriel – When You’re Falling (album version) – source: “Volume 3: Further In Time” (2001)

This one has the least definable reason for earning a place on the list. Of course, the production is stellar, and especially the recording quality of ACSS’s vocals has a clarity and presence that is unusual for a group of vocalists of that size. But for me, this is an exercise in raw emotional connection with a song that goes beyond any technical reason. I first heard/saw it used in an IMAX presentation called Adrenaline Rush: The Science of Risk, in 2002, and have loved it ever since.

2. Enya – Storms In Africa (Part II) – source: Bonus track on some versions of Watermark album, b-side of original 7″ single (1988)

Specifically part II, which has lyrics in English. The brightness of this version is unlike anything else released at the time. So many crystalline layers of treble in the vocals, strings, and percussion. The prominence of the African drumming in this version, coupled with the field recordings of rain and thunder give it a power that elevate it above the other popular Enya songs, in my opinion. I spent a long time (successfully) perfecting my own cloning of the arpeggiated synth sound that runs through it, and I spent hours playing that part by hand. This was one rare time where I specifically walked into a record store and bought a 45, just for the b-side. It became a “road trip tape standard” for years.

3. Frankie Goes To Hollywood – Welcome To The Pleasuredome (including all intro sections) source: Welcome To The Pleasuredome (1984)

This is a bit of a cheat, because I’m including “Well…”, “The World Is My Oyster”, and “Snatch Of Fury (Stay)” in this (which the link above misses), as I consider them all intro sections of the song. This is a flat-out masterpiece. Ostensibly a dance track, it has so many different things going on, it defies any genre pigeonhole. A perfect example of the determination and perfectionism of Trevor Horn, one of my personal idols for his contributions to the world of studio production. Every note and sound in this work was clearly painstakingly created and placed. Particularly noteworthy are the insanely complex bassline, backwards hi-hat loops, various layers of field recordings, brilliant guitar effect work, and sound palette encompassing so many different types of sounds (I tried counting once, and got lost after 100 ). This song never fails to make me feel awake and alive, no matter how I’m feeling before I listen to it.

4. Front 242 – Headhunter V1.0 source: Front By Front (1988)

Hey, I just discovered while looking for a link that “COMA Music Magazine ranked “Headhunter” as the greatest industrial song of all time in 2012.” Well ok then! Yes, I have the 4 CD set that has nothing but remixes of this song. But Version 1.0 is where it all started for me, and I had already been a pretty big Front 242 fan for at least a couple years before it was released. The way they proved that the Yamaha DX-7 synth, a staple of 80s synth pop, could be tortured to produce some really harsh sounds directly informed my own FM synthesis creation process for years. As a DJ at the time, putting this track on and watching the entire dance floor go ballistic was always deeply satisfying. Even 10-15 years later, I would still occasionally be in a club when this got played, and clearly there are a lot of people who still remember (or freshly discover) how perfect this song is for jumping around like a maniac. Also, I think I may have suffered a little bit of permanent hearing damage watching them perform it live during the Reboot tour, one of the best concert experiences of my life.

5. Imagine – John Lennon source: Imagine (1971)

What can you say about the quintessential song about world peace? Yes, I remember exactly where I was on the night of December 8th, 1980, and it had a profound impact on me. I could write a whole separate essay on how it changed my perception of the adult world, and how I felt connected to the adults in my life at the time because they were all as shocked and devastated as I was. I still stop what I’m doing on that day every year, and spend some time listening to John Lennon. I’m a decent keyboard player, and a decent singer, but I’m rarely coordinated enough to do both at the same time with any success. This song is one notable exception. If I ever had to perform live by myself somewhere, this is probably the only song I’d feel confident I could do some justice to.

Etherium Audio – Demo Scene ‘Zine article circa 2004

Here is an article that was published some time in 2004 for a Demo Scene ‘Zine online, and possibly even in print. I can’t remember exactly what it was, and can’t find it online now. If anybody knows, or knows a Lonnie Taylor (editor for the publication that I submitted this to), please let me know, as I’d love to read the whole issue again. I believe it also had some other contributions from fellow Northern Dragons members.

—————————-

After the release of the 4K demo Etherium that Northern Dragons entered into Assembly 2003, I got a few emails from people asking me for some details about how we accomplished the audio. The intention of this article is to provide a bit of insight into what we did, and hopefully answer some of those questions. You can check out the demo on Pouet at http://www.pouet.net/prod.php?which=10569, and visit Northern Dragons at http://www.northerndragons.ca

Our overall game plan, to leverage the skills of the people involved, was to create a complete sound engine module in C++, and have it converted to NASM and optimized afterwards. Since it seems that a lot of the DirectSound examples available online are in C++, it was a lot easier to figure out what we were doing that way. That, and I just felt a lot more comfortable doing a bunch of ugly trig in C++.

The playback engine is just a bunch of DirectSound intialization stuff that you get to by including DirectSound header files and libraries from .dll files, and defining some data structures. As a musician/math guy, it was nice to have a framework for all the initialization details created for me in C++, and then let the coders take what I had and put it into assembly. Taking that idea one step further, you can also use a framework like that to let different people work on the math, creating the sound palette, and someone else write the actual music. It’s all about division of labour, and making it a solid team effort.

The main thing I was working with was a huge ‘buffer’ in memory that you can think of as a wave. I don’t remember the numbers, but the track was about 2 minutes long, 16-bit (2 bytes) mono, at 22050 samples/second, so it’s roughly (2 bytes/sample)*(22050 samples/sec)*120 seconds =  5292000 bytes = ~5 MB. That’s a contiguous chunk of memory you can access like an array, and they’re all zeros to start with. When you’re done building into that, you just pass a pointer to it to the DirectSound engine, and say ‘here, play this’. In our case, we said ‘play this with stereo reverb’ (specifically, Direct Audio DSFXWavesReverb), so it sounds a little less dry. You can really notice it in the kick sound.

So then, you need your sounds. I built this whole thing with functions…calling functions…calling functions. The lowest level of function was just an equation for each sound. You pass it some stuff like a frequency, an amplitude, and a start position (your intial array index to write to the buffer). The equations are pretty basic trig functions, and I won’t really get into the details. They’re mostly one-liners (but ugly mothers) in a for loop. Believe it or not, the lead sound is almost a pure sine wave. The amplitude envelope is as simple as a quarter sine wave itself. Draw the quarter sine wave that goes from 1 down to 0, multiply the amplitude by that, and you get a really cheap non-linear roll-off. I also figured out how to do an inverse exponential roll-off just by taking the amplitude, and multiplying it by 0.999999 over and over, which I used on the whooshing sound.

One of the things I strongly urge people to do is use an initial volume ramp on their sounds. It’s the biggest thing that I hear other people missing in this sort of work, and it makes a huge difference. Those initial clicks you hear on every note in some productions…well, they can drive your audience crazy, and can’t be great for speakers either. Here’s a code snippet to show what I mean, and how easy this is:

if(count<=50)
value *= (amp/50*count);
else
value *= (amp*envelope_function(count)));

If you think about it, (50 samples)/(22050 samples/sec) works out to around 2 milliseconds. Enough to eliminate a click, but not enough to really damage your attack transients, unless you’re being really picky.

The kick drum was the cheapest thing ever. It’s a sin wave that drops from (I think) something like 60Hz down to 40Hz, with that quarter sine rolloff I was talking about. Dirt simple.

The whooshes were, as I said, generated with the random number generator. I just kept increasing the rate at which it chose new numbers for my samples, so the pitch of the noise goes up. The flanger took care of smoothing out the ‘burbling’ you would otherwise hear at the start. A real bonus here was the fact that we already had a random number generator for other areas of the production, so it turned out to have multiple uses, increasing our bang-per-byte.

The chords were a little trickier. I don’t even remember what the final version ended up being, but I think it was a sawtooth, with the points of the teeth flattened into something square-like, so it didn’t sound quite so harsh. The math for that, while not involving trig, was one of the hardest parts for me to get right.

A few words about effects. If you have one sound you want to put a static delay on, like our lead sound, it’s simple to incorporate the delay right into the sound, just by writing to 3 or 4 further offset locations in the sound buffer, with a scaled down amplitude for each. It makes the code pretty tight, and then you don’t need separate delay functions, or have to worry about using a DirectAudio delay effect that turns your whole mix into a jumbled mess.

I did write a separate flanger function, just because the code was pretty cheap and simple, got reused in a few places, and ended up being smaller than it would have been to turn on a DirectAudio flanger. When you’re building these effect functions, think hard about what parameters you want to control. My flanger had start and end times, depth, rate, and mix amount. Knowing which settings can change between calls, and which are constant can help a lot with optimization later, when every byte counts! Something else to think about for the conversion to assembly: parameters are cool, but more challenging for the folks coding it to assembly. Setting up the assembly routines to accept a pointer to a datastructure that holds all the values for a given effect is a cheap and effective way to create a simple music scripting technology. ’nuff said!

Then you have a melody block function that calls the sounds with the right frequencies, with a bunch of offsets (plus a master offset). Call that block a bunch of times in a loop, where the master offset increases, and you generate that melody block over and over. You build the entire thing with blocks built on blocks, and the whole thing becomes a lego exercise from that point. I think that part is where all the musical creativity goes, and I leave that as an exercise for the reader!

Hopefully that gives you some ideas, whether you’re considering doing this for the first time, or even if you’re a veteran in this area. Either way, I wish you tons of success, and look forward to having my socks knocked off!

Chris Deschenes (umdesch4)
Northern Dragons Audio Lead

Remastering Tips – Magic Man (addendum)

Nothing to do with the audio, per se, but after listening to this song SO many times, I noticed that nowhere online has anyone gotten the lyrics down correctly. Here’s what they actually are, according to my ears:

Magic Man – South Dakota (Before the Waves version)

Every time I stick around
It’s like you’re gone and we’re
Just the same and
Evidently we’re the smallest and I’ll
Get old
Get out of the summer

And…
Every time you come around
It’s all forgotten we
Never change well
Believe me it’s appalling you don’t
Get old
Get out of the
Space between the walls between the
Cracks under the floors won’t you

Rise South Dakota
Don’t you wanna know
You went so far now don’t you go home
and hide
Every time I run you run we run
I run you run we run, and we’ll

Rise South Dakota
Don’t you wanna know
You went so far now don’t you go home
and hide
Every time I run you run we run
I run you run we run

Remastering Tips – Magic Man Part 4

Alright, it’s been a long time coming, but I’ve finally found some time to get down to finishing touches.

First, I have a confession to make. I’ve been doing PC-based audio work for so many years that my workflow preferences are somewhat coloured by history. When I first started doing this stuff, computer horsepower was a serious impediment. You could only work with one bit of processing at a time, and never actually preview anything you were doing. Over time, this changed through successive stages, from realtime preview of an effect all the way to where we are now with the ability to stack tons of effects all over the place and hear fully realized renditions of them, even while tweaking parameters in realtime. I still haven’t fully embraced all that power, but there are exceptions…

Which brings me to my point, which is that remastering tool suites are awesome sometimes. There are a ton of plug-ins for different things, and at least as many opinions. Almost every discussion I’ve seen online includes mention of Izotope Ozone. If you’re serious about mastering, or remastering, it might be worth dropping some cash on it, or something like it. It can be used in any set of tools that support VST plugins, and I use it ALL the time.

Having said that, you can achieve similar (and maybe even better) results with individual effects, but I used Ozone for this part, so I’ll be showing it off a bit.

A further aside about things like Ozone. They come with stock presets for various things, like this:

Capture 12

The thing is, these presets are designed, at least partially, to be dramatic and show off how awesome the tool is, so when you see it demo’ed at your local music shop (upstairs at Tom Lee around these parts), you’ll be impressed and walk out with a boxed copy of the tool right on the spot. But if you actually use them as-is, you’ll end up with something that totally goes against my remastering philosophy of “subtle changes that improve things in a way that most people can’t quite put their finger on”. That’s not to say that they can’t make a great starting point for something good, which is in fact what I did with this project. You just have to dial things back (a lot).

Waveform pictures won’t make a difference at this point, since we’re messing mostly with EQ now, and not volume, so I’ll spare you a view of where we were at before now.

I actually started with a preset here called “Sparkle”. If you look at the picture above, you can see that there are six main sections to Ozone, and Sparkle is really just a combination of Paragraphic EQ (a hybrid between parametric and graphic EQs, but either would work for the simple tweak we’re doing here), and Multiband Harmonic Exciter. The other effects are all inactive.

So what exactly is a harmonic exciter? In a nutshell, it is a bit of black magic fakery. You take frequencies in your original source, artificially create integer multiples of those frequencies, and add them into the output. If, for example, you’ve got a pure note at 3500 Hz in your original source (roughly three octaves above concert A, for the musically inclined). You create some basic sine waves at 7000 Hz, 10500 Hz, 14000 Hz, 17,500 Hz. and add them to the signal. Sure it’s fake, but it gives you some higher frequency content that wasn’t there before. If you were only using EQ, sure you’d be boosting high frequencies, but if the high frequency content just isn’t there, it won’t accomplish much. Because the frequencies you’re adding with the exciter are harmonic multiples of something that does actually exist as a base frequency, you can be somewhat assured that the results will sound “musical” and won’t clash with the original signal.

Aside: Harmonic exciters are truly amazing. Even if you don’t cough up the cash for something like Ozone, you should get one if your audio editor of choice doesn’t have one. Freeware VST plugins are a google search away. They give you the ability to add that “sparkle” to just about anything. It’s especially important when remastering stuff from older sources where simply using an EQ to boost existing high frequencies results in the nasty side effect of also boosting tape hiss and other artifacts from analog sources. IMHO, no remastering toolkit is complete without some kind of harmonic exciter.

Alright, so using this stuff. I basically just pull the wave file I’ve got into Adobe Audition, go into multi-track mixing view (hit F12), drag the wave into the first track, click the fX button at the top of the track control column, and then select Ozone from the effects. Alright, that’s confusing…here’s what it looks like:

Capture 13

Now it’s just a matter of picking “Sparkle” from the list of presets, and dialing in the settings you want. As always, this comes down to listening carefully, and going for something that sounds good, and not over the top. Here are some screenshots that show you where I ended up:

Capture 14

Capture 15

You can also click on “Graph” on either of these views to show the order in which effects are chained. In this case it’s EQ first, and then the Exciter. As you can see, I did what I could with EQ settings to gently boost the high frequencies that were there first, and then applied the Exciter. Only band 2 does anything for the exciter, which uses frequencies in the range of 5.38 kHz to 20 kHz for the multiplier I was talking about before. I can’t remember what the amount was in the original preset, but I dialed it back to 1.6, and that seemed reasonable to my ears.

After you’ve got a setting that works, you need to mix down the results to a file. In Adobe Audition, this is under Edit->Mixdown to New File->Master output in session (stereo). The resulting file will look a little like the one below (on which I’ve highlighted a couple of areas for the next section):

Capture 16

This sounds great, but there are two problems, and they’re both in the red boxes above. Unsurprisingly, they’re both a result of the last step adding too much high frequency content to the the sibilant “S” sounds at the start of both instances of “South Dakota” in the vocals. In addition to being too loud compared to their surroundings, these two minor peaks keep us from normalizing the entire track to a more suitable volume. Rather than attempt some kind of compression, we’re going to spot-fix these.

This is where it’s again important that you really learn the navigation controls for your wave editor of choice. I’ll show one of the two fixes here, but it’s pretty much the same for all minor spot fixes to errant peaks. In this case, I’ll use the second, because it’s much more obvious. First you zoom right in on the part you want to fix, and select it. It will be something like this:

Capture 17

As you can see, this is barely two tenths of a second that is really at issue. As long as you only drop this by as little as absolutely necessary, it won’t even register much to the listener, and make a big difference overall.

For this kind of thing, you’ll want to use envelope processing. In  Adobe Audition, this is in the amplitude section under effects. I played around with an inverted bell curve to get what I wanted. Make sure you have spline curves checked so you get a smooth transition. Here’s what mine looks like:

Capture 18

Here’s what that same peak looks like after processing:

Capture 19

As you can see, the peaks have been brought into line with the peaks in the adjoining audio. While auditioning this including a few seconds on either side, it’s barely noticeable. You might need a fair bit of trial and error to get the ramp amount exactly right, but it’s pretty quick to undo and redo this a few times to get it perfect.

So that’s really about it. After you’ve adjusted the peaks, you can normalize the whole thing as we’ve done in previous steps, and get a final waveform. In my case, here’s the final result after all the final fixes, and normalization:

Capture 20

Here again is the final result:
https://dl.dropboxusercontent.com/u/4006268/South_Dakota/6.ramp,norm,hammer,sparkle,spot-env,re-norm.flac

I hope this has served as a somewhat valuable tutorial. A lot of the ideas in here are mix-and-match, and should hopefully be useful for attacking specific challenges, even if what you’re doing is completely different.

That’s it for this project, but feel free to leave comments for things I’ve missed, been wrong about, or that need more clarification.

I’ll be back soon with more. It’s rare that I go for more than a few weeks at a time without some kind of audio project in the works, and I’ll be sure to write about whatever I’m working on next. At the moment, I’m messing around with Unity, making 3D environments to play around with using my Oculus Rift development kit…but I’ll get back to audio soon, like I always do!

Cheers,
umdesch4.

Addendum: https://umdesch4.wordpress.com/2014/12/24/remastering-tips-magic-man-addendum/

Remastering Tips – Magic Man Part 3

Now we’re going to get into more dangerous waters. Bringing up the subject of dynamics compression at a dinner party of audio engineers is like bringing up Mac vs. PC with computer geeks, or XBox vs. Playstation with console gamers.

As I said before, the argument stems from over-use. In my opinion, compression is always needed on a mix, unless it’s incredibly well balanced to begin with, contains a good amount of compression on the individual track recordings, or came from all-digital source instruments. Even then, you’ll probably want to compress somewhat, unless you know your target audience is going to be sitting in a dead silent room, on a couch in front of an expensive stereo system with a nice glass of scotch in hand, quietly focused entirely on listening to the music. And who (besides me) does that?

Ok, so now I’ll take a deep breath, and try to describe what a compressor does. I find that all the technical explanation in the world doesn’t help a lot with using compression, but at least if you know the idea, it can help with figuring out which settings to mess with if you’re not liking what you hear. Compression basically squeezes out the volume differences in a recording. Think of it like a robot hand on the volume knob that is able to quickly turn it down when the sound gets too loud, and back up again when it drops. I’ve always liked this analogy, so I’m going to stick with it. There are two main settings on every compressor, threshold and ratio. The threshold is the volume, in dB, at which the robot hand should start paying attention. Everything below that level should remain unaffected. The ratio tells the robot how hard to twist the volume knob in response to signals above the threshold. As implied, it is a ratio of input to output, so if your ratio is, say, 2:1, it will turn things down enough to make the output only half as much “louder” than the threshold as the input signal. Expressed another way, if a signal goes 2 dB over the threshold, the volume will get turned down to bring it to only 1 dB over the threshold. The higher you set this ratio, the more “squashed” your output will be. In fact, another type of dynamics control called a limiter is really just a compressor with an infinity:1 ratio. The volume gets turned down so that no matter how high the input signal becomes, it’s always turned down enough to not exceed the threshold.

Other features of a compressor…Attack and release times: Briefly, these tell you how quickly the volume knob gets turned in response to volume changes. Peak vs. RMS sensing: This determines whether the robot is reacting to individual spikes in the waveform, or root-mean-square averaging of the signal that is more indicative of “power”. This is useful if your track has a lot of spikes in it, and you find the compressor overreacting in response to a lot of attack transients. You’ll know it when you hear it, trust me. I usually use RMS sensing when remastering, because I know that RMS more closely aligns with human perception of volume, so the result is more natural-sounding, and there aren’t usually any errant transient peaks in something that’s already been mastered that would need fixing. Make-up gain: What we’re talking about here with compressors is technically referred to as “downward compression”. We’re only ever correcting the volume downwards in response to higher input signals. (Upward compression is a whole different thing, done by an expander, and that’s a whole separate discussion) Since we’re turning things down all the time, make-up gain is like a second volume knob that stays fixed at a certain increased amount as overall compensation for the volume reduction the compressor is doing. Look-ahead: Our robot is quickly reacting to input signal as it comes, and making all these volume adjustments for us. But it can only deal with signals once they’ve come in. Real world hardware compressors must work like this, because they can’t see into the future. There are tricks in hardware that can overcome this, which are a little like live network television broadcast censors. They can bleep out all the ‘fucks’ as they come in, but the entire broadcast ends up being delayed a few seconds to allow this to happen. But in digital editing of an already recorded track, we know the future, so we can let software compressor in on it. Then it can react right at a sample level, and not let those transient peaks through because it couldn’t react fast enough. ’nuff said.

Whew. Ok. So let’s do this. Here again is the waveform we’re working with:

Capture 8

At this point, I will state a compressor preference. Sony’s Wave Hammer is like magic for me, and I’ve been using it since somewhere around 2001. In fact, it is pretty much the only reason I keep Sound Forge installed at all times. It has some super-secret-sauce features beyond what a regular compressor has, and a healthy set of presets for dealing with individual instrument tracks, and mastering. I like to fire the thing up, and start with the “Smooth Compression” preset. Here’s what it looks like in action:

Capture 9

This is a capture of it while previewing. I just wanted to point out that the red bar on the right side shows you how much the compressor is actually clamping down on the input signal. All software compressors that you can preview with include a display like this to show you in realtime what they’re actually doing.

So this is a good starting point, but now there’s a bit of tweaking to do. This involves a lot of listening, making slight adjustments, listening again, etc. As I said before, I like RMS sensing better for this kind of thing (Scan Mode: RMS in this case), and I found that the threshold and ratio were a little too aggressive, so I ended up adjusting them. I also checked “Use longer look-ahead”, although I honestly didn’t hear much of a difference.

The “secret sauce” I referred to seems to be a combination of ‘Auto gain compensate’ and ‘Smooth saturation’. You’ll notice that Output gain (which would be the make-up gain I talked about earlier) is set to zero. But still, some amount of volume increase happens to the quieter parts of the track. By the way, this has nothing to do with the Volume Maximizer tab that I’m not showing here. In this preset, it’s all zeroed, and you can bypass it entirely with no effect whatsoever. Anyway, whatever the authors of this tool have done, I’m loving what I hear, so here are the settings I chose before hitting ok:

Capture 10

Here’s what the resulting waveform looked like:

Capture 11

You can see that it doesn’t look like a huge difference, and in fact it really isn’t. Again with the subtle adjustments philosophy. But every section of the track is slightly higher overall, and fatter. It sounds like it too. Here’s the output, if you want to give it a critical listen: https://dl.dropboxusercontent.com/u/4006268/South_Dakota/4.Volume ramp(normalized)(slight hammer).flac

Sorry, no sparkle yet. That’ll be next time…

Cheers!
umdesch4

Part 4 here: https://umdesch4.wordpress.com/2014/11/11/remastering-tips-magic-man-part-4/

Remastering Tips – Magic Man Part 2

Alright, so now we have a solid clip to work with. Let’s pull it up again and take a look:

Capture 3

At this point, what I always do when I’m either remastering something old, transferred from a non-digital source, or simply mastering something I made, is normalize it. This is so basic that many editors don’t consider it an effect at all, so you might find it under your Edit menu, or in the case of Sound Forge, under the Process menu. (Audacity puts it under the Effect menu, and Adobe Audition puts it under the Amplitude and Compression section of the Effects menu)

Normalization is a dirt simple process that simply scans the whole waveform, finds the highest peak, figures out a multiplier that will take that peak to the maximum value that can be represented by your waveform’s bit depth, and multiplies every sample by that amount. Too fancy sounding? In real terms, it cranks the master volume knob up to the maximum it can go without causing any distortion. No matter what you’re doing during a mastering process, you want to start from this point, because you won’t be mixing tracks together so you won’t need any headroom, and you want to push everything to the best  workable volume before you start manipulating anything.

Now that I’ve said all this, I have to point out that in this example, starting with normalization does nothing for us. In general, this is true whenever you’re working with already produced digital sources. You will almost never come across a commercially available recording that isn’t already normalized. If you look again at the waveform, you’ll see that near the end of it, there are individual peaks that pretty much max out the waveform’s boundaries, so there’s no room to boost the volume on this without those peaks getting clipped.

This leads us directly to one of the main problems I have with this recording, and a brief interlude…

Brief Interlude – Mastering/Remastering Strategy

Whether you’re mastering new material, or remastering existing stuff, you have to have a plan. It isn’t enough to just say “I want it to sound better”. You have to be able to determine the issues and limitations of the recording you’re working with, and figure out what you want to achieve. If you really don’t know, it can help to compare it to an existing recording that sounds like what you want, and figure out the differences. Things like “it’s generally louder”, “it has punchier bass”, or “it’s brighter and more detailed”.

In this case, I knew exactly what my problems were with the original. The first was that the dynamic range was too great. The difference between the quietest and loudest portions of the song are so dramatic that you need to adjust the volume over the course of the song to hear it properly, or else you either get annoyed that you can’t make out the beginning, or you blow your eardrums out near the end. As an aside, I understand the artistic direction they were going for here. It’s a “hidden” bonus track, so they wanted it to sneak up on you, and they wanted to build it up dramatically to overwhelm the listener. Sure, that’s great when you first discover the track, but I’ve been listening to it on repeat for months, so the novelty of that wore off right away, and I just want to listen to it without fiddling, while still preserving some of that effect.

The second issue I have is that it lacks clarity. Mostly in the high end. There is some content there to work with, thankfully, but overall the mix in the first half of the track is so dull sounding that I’d like to make the vocals and metal of the drum kit clearer. That’s entirely subjective. I just wanted it that way, because I generally like my recordings to sound crisp, clear, and detailed.

Those are the two things I was looking for, and I’ll give a piece of universal advice here. The order in which you do things is actually important. As a rule of thumb, which I’ve learned for myself, and other engineers have also confided about, you always want to do any volume or dynamics processing BEFORE you take on EQing issues. This is because odd things can happen when you’re changing the volume of a signal, especially using dynamics compression, that will change the character of the sound. If you EQ before that, you may find that you’ve overdone something, and have to go back, or that you’ve somehow defeated that you were trying to accomplish, and need to do more EQing after. The fewer steps in your process, the better, right?

So this is why the first thing I’m tackling is the volume issue. Here we go…

Volume Shaping With Clip Envelopes

I’ll admit it, I almost always throw a compressor on things right away. Yes, if you google “loudness war” you will find a ton of digital ink spilled about it. I agree with everything they say. But like anything, it is a case of “too much of a good thing” (and for non-artistic motivations) that has ruined so many recordings in the last few years. Compressors/limiters with gain compensation can achieve incredible things though, when used properly, and I’ll be talking about that next time.

The reason I’m not talking about it THIS time, is because I have to confess that I spent at least 45 minutes with this track throwing every dynamics compression tool at it I could think of, and nothing sounded good (yet). I’ll spare you the details, but I quickly realized that I had to do something much more basic and global to the track before I could get to that degree of volume tuning.

So the real problematic portion of this track is from around 1:48 to about 2:37. You can see what they’ve done just by looking at the waveform. The volume steadily increases, in an almost exactly linear fashion. It’s a fairly simple matter to “undo” this, and see if it helps.

For this, the easiest way is to pull the waveform into a multitrack view. Don’t ask me why most editors don’t let you do this easily in a single wave edit view, but it’s generally true. My preferred editor for this is Adobe Audition, but most editors work pretty much the same as this. The feature we want is almost universally called “clip envelopes”. To see and/or edit them, you may have to find setting in your pulldown menus that enable them. Audition puts all those under the view submenu, like this:

Capture 4

In Audition, you have some additional fancy features for working with envelopes (under Clip->Clip Envelopes submenu) that let you create smooth splines. For this, I really don’t need that. Straight lines work fine for me!

Once you’ve enabled these things, you’ll have the ability to graphically manipulate the volume (and pan, if you want to) of the track over time. Every tool I’ve ever used like this puts the volume line at the top, and the pan in the middle. In Audition, the green line at the top is the volume envelope we want to work with. To start, it simply looks like this:

Capture 5

The tiny white boxes at the upper left and right edges of the picture here are control points that you can click and drag to move this line around. As you’d expect, dragging down will lower the volume at the control point, and you’ll end up with a diagonal line connecting the two points. Fairly intuitive, once you start doing it. As an added bonus, when you hover over or drag a control point, a pop-up will show you the value, in dB, for that point. At +0 dB, you’re at full volume, and dragging down will bring you into the negatives.

Aside: dB, or decibels, are weird man! So is human hearing and our perception of “loudness”. Without getting too far into it, an increase or decrease of 3 dB is technically a doubling (or halving) of power, but it takes a 6dB change to result in a perceived doubling (or halving) of volume.

Where was I? Oh yeah, clip envelopes, control points…right. So you can create any control points you want just by clicking anywhere on the envelope line that doesn’t already have one. Drag ’em around, and make any kinds of volume adjustments you want. I ramped down the volume between 1:48 and 2:37, and then fairly quickly brought it back up again to full immediately after the last explosive drum hit. This involved a fair bit of zooming, panning, scrolling around, and auditioning the changes to make sure it sounded right, so this can take longer than you think. I settled on a drop of 5dB at the lowest point, because it ended up sounding reasonable, and flattened that section of the track almost perfectly. Here’s a zoom-in of what my envelop looked like:

Capture 6

When you’re working in multi-track, you typically have to “mix down” the results to get an output waveform. In Audition, it’s under file->Export->Audio Mix Down. This will save out the envelope changes to a new file. Pulling that file back into a regular wave editor, here’s what I got:

Capture 7

Pretty sweet. It still sounds good to my ears, and I bought myself nearly 6dB of headroom to work with. Hell, I’m going to normalize this right away! As I described above, that will maximize the volume…but now, it will effectively double the volume of the entire first section of the track, and bring the very loudest parts back up to roughly where they were before. That’s a simple matter of finding your normalize function, picking 100%, applying it, and saving out the result. Here’s what mine looked like:

Capture 8

That’s a lot of progress, IMHO. I’m not nearly done, but this is a good point to take a critical listen:
https://dl.dropboxusercontent.com/u/4006268/South_Dakota/3.Volume ramp(normalized).flac

…and about critical listening, some more advice. These kinds of changes can be very subtle. In fact, that’s often what you’re going for with remastering – making changes that improve things in a way that most people can’t quite put their finger on. So listen carefully, and try not to look at what all the waveforms, envelopes, knobs, and sliders are saying you should hear. Don’t even look at the screen! And try not to think too much about what changes you’ve made. Also remember that music listening is an emotional experience. The track you started with must have inspired you in some way that made you want to work so hard on it (unless you’re getting paid), right? If what you’re hearing makes you smile (or want to bawl your eyes out, or jump around the room like a madman) as much, or hopefully more than what you started with, you’re going in the right direction.

In the case of this track, you might notice that even though the entire last section looks like it’s all at the same volume, it still seems like it’s building and getting more powerful. There are other tricks the band has pulled to make it sound that way, and messing with the volume doesn’t seem to have diminished this. At this point, I’m still loving the track as much as I ever did, and now I can hear it all just a bit better, so I call this a win so far.

Next time, more dynamics tweaking with compressors, and maybe some sparkle.

Cheers!
umdesch4.

Part 3 here: https://umdesch4.wordpress.com/2014/09/21/remastering-tips-magic-man-part-3/

Remastering tips – Magic Man hidden bonus track example

Background

The hidden bonus track at the end of Magic Man’s 2014 release Before the Waves is a reworking of “South Dakota” which originally appeared on Real Life Color, released in 2010.

The new version is beautiful, but difficult to listen to because of it’s extreme dynamic range, and slightly muddy EQing. Because I’ve been listening to this track endlessly for months now, I decided to do something to make this track sound as great as it deserves to sound, so I could listen to it myself without the mixing issues affecting my enjoyment of it.

If you want to hear it, albeit stripped of some of it’s glory via the anti-miracle of lossy mp3 compression, you can check it out at https://soundcloud.com/umdesch4/magic-man-louth-dakota-umdesch4-remaster

Edit: Here’s a FLAC, so you can hear it properly, and decide if I did a good job or not – https://dl.dropboxusercontent.com/u/4006268/South_Dakota/Magic Man – South Dakota (umdesch4 remaster).flac

Since this actually turned out to be a lot of work, I decided it might be worthwhile to write a tutorial about what I did.

Part 1 – Prepping the track

This is admittedly dull stuff, but even here, there are some things to talk about that I feel worth sharing.

First and most obvious, you have to choose an audio editor in which you’re comfortable with the basic operations of scrolling around a waveform, zooming in and out, and standard cut/copy/paste operations.

I generally use two tools to do the bulk of my work: Sony’s Sound Forge Audio Studio, and Adobe Audition. In general, I find Sound Forge easier to use for extremely basic stuff like this, without too much interface clutter. Anyway, whatever you’re comfortable with should be good enough for this kind of thing.

So, here’s what I started with. You can see that there’s the “regular” part of the track (track 12, It All Starts Here), followed by a big chunk of silence, ending with the track I want to work with:

Capture 1

Rough cropping of this is fairly straightforward, but this particular example presents a bit of a challenge (not unlike many other things I’ve dealt with in the past). Where are the actual start and ending points of the bonus track? It’s tricky because there’s a fade in, and the overall volume of the intro is so extremely low that it’s hard to tell exactly. You can argue that it is simply a matter of zooming in to where you think the beginning section of the track is, cranking your monitor volume, and listening for it. That may not be good enough. Especially when dealing with 24 bit samples, it may turn out that the noise floor of the whole output chain to your ears is higher than the actual signal. This wouldn’t matter much now, but if what you intend to do later involves some heavy boosting (normalization, dynamics processing, whatever), it may turn out that the final result magically rises about this noise floor, and now you get to hear exactly how you’ve missed the start of the track. Whoops!

In this case, it turned out that I could hear the actual start (more on that in a minute), but just in case, you can do a visual inspection too. The simple trick is to zoom in on the area where you think it is, and then zoom vertically (ie. amplitude-wise) as far as your editor will allow you to go. In Sound Forge it ends up looking like this:

Capture 2

In Sound Forge, at least, the +/- buttons on the far left side are what you spam to achieve this view. The +/- buttons on the far right are for zooming in and out in time. (Sorry if that’s obvious, but hey…)

So now you can clearly see that there’s definitely signal at around 5:38. Listening to this at a fairly high volume, it seems like the musical fade in starts much closer to 5:40. Indeed it does, but there’s something interesting going on during those first 2 seconds. The band decided to start the recording a little early, and if you seriously crank it, you can hear a 60-cycle hum coming from their guitar amp. It’s subtle, but it is (IMHO) a powerful subliminal cue that sets up the whole character of the kind of recording you’re about to hear.

What you want to do with this is crop your recording, keeping roughly a half-second before this point. The reason for this is that, especially with advanced dynamics processors, there is often an option for “look-ahead” processing, and you want to give it at least a few extra handfulls of sample-space to work with. Also, caution here is a good thing, as you can always crop more after you’ve done everything else.

For the end point, you can use the same techniques. The difference is that you want to leave even more room at the end. It may be that you feel the need to introduce some slight mastering reverb to the source, and you will need room for the reverb tail. Also (although not as important here), one component of aural enhancers/exciters involves time delay of specific frequency components, so if you’re going to get extreme with some of these exotic effects, you’ll need the room. If the original recording doesn’t have room for these tails, you may want to cursor to the end of the selection and use whatever “insert silence” utility your editor has to drop an extra second or two in there.

So yeah, that’s the easy part, but getting it wrong can lead to headaches later on, so take a few extra minutes doing it right. Measure twice, cut once, right? Once that’s done, save out the cropped results to a new file, and you’re good to start the real work.

Here’s a FLAC file with my results. It’s a handy reference to the original, so you can judge for yourself how well I did at the various stages in the next sections:
https://dl.dropboxusercontent.com/u/4006268/South_Dakota/1.Magic Man – South Dakota (unprocessed).flac

Part 2 coming soon!  (here: https://umdesch4.wordpress.com/2014/09/17/remastering-tips-magic-man-part-2/ )

umdesch4

Introduction

Hi, I’m umdesch4, and I’ve been a hobbyist audio engineer for over 35 years, ever since the age of 4 when I figured out how to thread an open reel tape machine, and hit the red button. This was much to the chagrin of my parents, when they came into the room and discovered I had managed to erase a significant chunk of one of their good reels. Oh well…

Through the years, I’ve done all kinds of crazy experiments, making tape loop collages from dissected cassettes, chaining together tape decks to painstakingly achieve some tape delay effects, and do very rudimentary overdub recording. Once I got my hands on a computer (the first serious one with musical possibilities being the Commodore 64), I incorporated that into my toolkit. I’ve written SID chip compositions, later MODs on the Commodore Amiga, and began messing around with digital sampling. Also being a musician, I got deep into MIDI sequencing, at various points writing my own bits of software to accomplish various things.

I never let the real-life aspect of things slide either. Around that era, in the late 80s, I was also a DJ, and went the extra mile to make sure everything sounded good, and there were racks of lights with chase and strobe patterns linked up to the music I was playing.

Since those days, I’ve taken audio engineering courses, so I know my way around mic placement for a drum kit, studio multitrack recording, various automation systems, and I’ve helped bands record demos, done some ADR and foley work, even multi-track recorded a string quartet in an apartment with double mics on each instrument and an array of room mics too! I’ve done live sound for bands at local festivals, and done a lot of my own field recording. For every situation where the average person would be taking video or pictures with their mobile gear, you’ll probably find me with a “prosumer” digital audio recorder taking high resolution surround sound audio field recordings.

Oh, did I mention, I’ve also done a lot of surround sound recording, composing, and mixing? I started by developing my own “poor man’s surround” where I reverse-engineered Dolby Pro-Logic Surround, and injected my own filtered sounds into a stereo mix to produce the same effect. Then I figured out how to do true surround sound, and master it to DVD-A, which is still my preferred format.

Anyway, it seems like everything that I learn in my life…physics, math, electronics, programming…I try to relate back to how it can be applied to audio, so that’s very much a part of the kind of person I am.

Of course, I spend the bulk of my time these days doing various little audio tasks for people, and myself, entirely in software on a PC. That’s mostly what the next few posts in this blog are going to be about.

I hope some of the things I’ll be talking about are informative and interesting

Cheers!
umdesch4

umdesch4