Digital Format Tiers

So a while ago I created this hierarchy for different kinds of digital audio formats. I've wanted to stay away from audio quality and format related posts, but I've been neglecting this blog... and sometimes I wonder whether or not I have anything else to say! This rating is intended to be a way of finding out how much of an audiophile you are. What tier you are on is determined by the quality of the majority of the audio files you listen to. Here are the tiers:

Tier 0: -Less than CD-Quality, meaning 192Kbs or lower MP3 or other compressed formats, this is the minimum for listenable audio reproduction. An extreme example of this would be the low-bit movement, which is actually pretty cool.

Tier 1: -CD quality WAV or FLAC, and ~320kbs MP3/M4A/OGG for mobile listening.

Tier 1.5: -Approximately CD quality sample rate (44.1/48K) but 24 bit resolution.

Tier 2: -Higher than CD quality, 88.2/96K WAV/FLAC, always 24 bit.

Tier 3: -Stupidity. 176.4/196K sample rate, 24 bit or 32 bit float.

Tier 4: -Vinyl + magic turntables, solid gold cables, speaker crystals, etc.

The last one is obviously a joke, vinyl and tape are a different beast altogether from digital audio. But it is true, you can waste a stupid amount of money on obscure and expensive stereo equipment that is supposedly better than even the highest resolution digital system. That's up to your ears, and mainly your wallet whether or not you want to buy into it.

Tier 0 is serious though, I admittedly still have some 192kbs MP3s. They don't even sound that bad because of how well MP3 and other codecs can compress the audio data. One time I found out that a favourite album, which shall remain unnamed, was encoded as a 64kbs MP3. That's the amount of data is takes to make a telephone call! But to accurately represent a CD quality file, it takes a 192kbs or higher bitrate MP3. I use M4A myself, on one hand because I did a shootout and found it to be the closest to representing a CD quality WAV. But on the other hand, it's what iTunes uses by default, so you can chuck your CDs in your computer and it converts it nicely with the files all nicely tagged. (Minus a million geek points)

Tier 1 is "legit tier" meaning the highest that consumer audio is delivered at, as well as the lowest quality that is acceptable for professional music production. This of course means 44.1K/16-Bit, stereo WAV files. These are fine for production, if you are into sampling than you probably know the history of CD quality sampling. There were some early, much sought after samplers that sampled or stored data at slighter lower than CD quality, but that's beyond the scope of this post. The most sampled drum break of all time, the amen break, is widely available online as a mono CD quality WAV, ready for chopping up.

320kbs is the maximum bitrate that the MP3 format specifies. You can make MP3s at a 48khz sampling rate, but it's not really recommended. Firstly, they might not play in certain players, or they might get converted down to 44.1K anyways, probably by a dodgy algorithm. So you'll lose the 8.125% extra quality you've added (o no!) But also, you're increasing the sample rate while the bitrate is still the same, so an equal amount of data needs to represent more samples, thus lowering the quality to the same level. Basically, if you want to get the most out of your MP3s, you should adhere to the 44.1Khz/320kbs standard.

For other formats, you can get a bit higher than 320kbs, but your filesize will be slightly larger. OGG, a newer but very widely supported file format, can go up to ~500kbs, but I find it difficult to find reliable software to convert files to this format. It's not worth the extra hassle and disc space to get the most out of compressed audio. Maybe one day things will be better, and we'll all be using 500kbs OGG at 48khz on our future smartphones. Sounds utopian, which means it will probably never happen.

Tier 1.5 I added because there's a crossover, as well as a sweet spot in this area. It's for audio at 44.1/48K but at 24 bit resolution. For most people, it's not really necessary to have audio at any higher quality than this. You can rip a vinyl at this rate, FLAC it, and it will be the same size as a CD- quality file. That's what I like most about this area, you can have your cake and eat it too, and you won't get fat either! Well, your disks won't anyways. (Get it! FAT, disks! *cough*cough*) Anyways, I always recommend recording at 24 bit, even if you're using a single resolution sample rate like 44.1/48K. It will make life easier for you as a producer/mixer, if you're into that sortof thing.

Tier 2 is studio-quality. Most studios these days record at 96K/24-Bit, or even higher rates. I think it's overkill for a number of reasons, some more justified than others... But, this is what the pros do, so if you really want to be like them, than use double resolution sample rates like 88.2/96K and always 24 bit resolution. But you should probably have an external drive for your sessions, becuse all those tracks of high quality WAV/AIFF files are going to clog your drives up. A 88.2k/24-Bit file is 3x the amount of data as a CD quality file, per mono track. This is the main downside to using such a high resolution format. But chances are your interface runs at 96/192K, 24-bit, so if you really want to get the most out of your gear, than you should use this rate.

Tier 3 is where you get into serious overkill. A 192K/24-bit file is 6.5 times as large as a CD quality file. Totally stupid in my opinion, and in the opinion of alot of professional audio engineers. Yet for some reason alot of equipment runs at this rate, for what reason I'll never know. Everything I have read on the matter has been to the effect that 192Khz is an unnecessarily high sample rate. This is one of those cases where bigger does not = better. 96Khz is all that is nessecary for extremely high quality audio, higher than you'll ever be able to perceive with your own (likely damaged) hearing.

I have a personal anecdote to explain Tier 4. The singer in my Dad's band works at a company that sells serious audiophile-grade gear. The last time I saw him, he said they had a turntable that cost $100000! There was also a pair of cables that cost $15 grand. When I asked "Who would buy that?" the answer was of course "People with way too much money. They say 'Oh that's the best? I'll have that then.'" So, with that in mind... What would you do with $115 grand? I would spend 1% of that on a Technics SL-1200, than buy a bunch of vinyl. Than I'd put a down-payment on a house to put it all in, than buy a car to bring be back and forth from DJ gigs. Sounds utopian right?

Probably because it will never happen.

Until than, I'm keeping my headphones on.

Creative Blocks/Ego

So I've been staring at the blank page, or more accurately DAW project for years now. It's hard to look at yourself in the mirror and accurately evaluate your own flaws, but that's what's needed in order to get past a block. With this blog I've laid out everything I know about modern recording technology. But I feel like I haven't said enough about music, well that's because sometimes music puzzles me. Why some sounds are accepted and some are not, why some combinations of notes sound pleasant to one person and unpleasant to another? It seems hopelessly subjective, but there are some things that are generally agreed upon. One that is very hard for me to accept is that analog distortion tends to sound a hell of alot better than digital distortion! They're completely different animals, but a vast majority of people will recoil in horror when they hear digital distortion.

Personally, I don't mind it. I wouldn't care one little bit if a radio was tuned to static rather than pop music. I prefer the white and pink noise cascade of an untuned radio over most modern pop music. But I'm starting to understand the appeal to pop music, it's pleasant and carefree attitude. Often the production behind mainstream artists is more of an artform than the actual artist themselves. This is something I can get behind, but I don't think I'll ever understand why everything needs to sound so... similar. To the point where it doesn't seem to matter which artist is playing, you can hear ten songs in a radio playlist and they all sound like one meta-band jamming to a single vibe. It's some kind of groupthink, mob-mentality, and I think there might even be a touch of black magic behind controlling this force. But hey, as long as it's a good vibe...

I have to accept that my music will always be a minority thing.

The problems this implies are multiple in nature. For one, when you set out to make a noise track, who exactly are you pleasing other than yourself? How do you know when you're finished? I assume the answers to these questions are as personal as they are for any other kind of artist. Maybe someone can help me fill this part in. But this leads me to the heart of one of the major dualities I've discovered while thinking and writing about music: Representationalism vs. non-representationalism.

Big words, scary shit. But in the context of recording, it comes down to this: When you record a sound, say an acoustic guitar, do you want your end product to sound like an acoustic guitar? If yes, than this is an example of a representational recording. It's like drawing a portrait, as opposed to a Picasso painting. Now, say you sample a string on an acoustic guitar, and pitch it down until it's a bass note, than you apply some kind of filter effect and modulate it, this is an example of non-representational recording

The problem for me is that most pop music makes use of pitch correction, heavy editing, and other effects that make the recording less than representational of the performances. My main point is that as technology has gotten more and more advanced, the palette of music that is commonly available is getting more and more homogenous rather than less. If you have these incredibly expensive studios that can transform almost anyone's performance into a polished sounding, radio-ready recording... why aren’t they being used to turn the sound of breaking glass into music?

I suppose I'm just as vain as pop musicians in many ways. There's no real need for anyone to be perfect, to try follow my production advice to the T. It's just MY ideas, some arbitrary rules that should theoretically lead to a more ideal sound. Yet perfection doesn't really exist, it's an abstract ideal nobody really reaches. So why try? I can think of a few reasons but it's hard for me to care anymore. I feel like giving up on being a conventional musician, and start crafting noise into sound-art. This seems depressing to some people, but for me it's incredibly liberating. But I'll never stop playing, practicing, jamming with other people, trying for the sake of trying.

There is a social element to music, to playing in bands, to "trying to make it big" or even pretending you are for the duration of your set. This is maybe the only non-selfish, non-narcissistic aspect of music I can think of. Even though from the outside it might seem like the essence of egotism, narcissism, to think you are a rockstar when you're not. The point though is that it's not real, but to a certain degree you need to pretend like the show is real. The ambiguity is what makes it all possible, just don't get lost in it.

Link Dump/Clipping Discussion

Here's some more beating of a dead horse, I saw this on the blog http://junglistmanifesto.blogspot.ca/

Also thanks to Junglist Manifesto for their Jungle Jungle sample pack. Great shit.

Here's a great guide to levels in digital audio, which also goes over 32 bit audio, as well as the effects of digital clipping from PopMusic.dk:

http://www.popmusic.dk/download/pdf/levels-in-digital-audio.pdf

Lately I've been experimenting with limiting, and intentionally clipping a mix I did with a friend a while ago. It has been fun, and interesting seeing how much volume I could add to an already compressed mix by limiting it. When I was first mixing I was always frustrated that I could never get the volume, or perceived volume that commercial tracks seem to have. So I got some free diagnostic tools, looked at my favorite mp3's and discovered that a vast majority of them are clipping, sometimes VERY hard. I noticed some electronic music clips at +6dbs, which basically means the top 6dbs of audio are flattened, but the rest of it sounds TWICE as loud as a file which has been normalized to 0dbs. What's going on here?

Well, you can actually clip somewhat hard and still end up with a listenable mix. You just sacrifice the peaks of your audio, and get some artifacts from the process. The problem is that the dynamic range of the audio is reduced. This can result in a "flat", "harsh" or "loud" sound being permanently imprinted on the mix. Again, this is one of those processes that you can't go back from. Once something has been exported as a fixed point (16 or 24 bit) WAV file that is clipping-those peaks are gone. So once again, save copies of your projects! Clipping isn't really something you should be trying at home, unless your a mad scientist who enjoys brutalizing the ears of your listeners! Check out this Mixdown for Mastering guide if you are curious about how to export your mixdown for the mastering stage.

Audio Quality Wrap-up

So I've decided to sum up everything I know about audio quality in order to make a final, definitive statement and put the topic to rest. What I've found is that CD quality audio (sound sampled at 44.1Khz/16 Bit) is perfectly adequate for 90% of uses, and is even overkill in some cases, such as recording a very lo-fi source. It thus follows that a vast majority of samples, recordings, etc are in this form. When making music digitally we will most of the time be dealing with audio in this format. So we of course need a way of dealing with it, and incorporating it into sessions with higher quality audio.

Another basic point is that everytime you convert your audio to another sample rate, or bit depth, you are reducing it's quality. My earlier posts were simply pointing out ways of making the reduction in quality as minimal as possible, preferably pushing it down to inaudible levels. It is true that most of the gradations of quality I talk about are close, or very close to inaudible through the average consumer stereo or speakers. So if you don't really care about hi-fidelity listening, don't want to think about how something sounds through even average studio monitors or medium quality headphones, than by all means use CD quality and go outside, play a round of golf, read a book...

OK, I'll assume only the hardcore audiophiles are left. So, what is the best workaround for CD quality audio? Well, if something is already in this form, I don't really recommend upsampling it, because you'll have to convert it back down anyways, and you should ideally do as few conversions as possible. All you can do is save any audio you export as a 32 bit float to make sure that all the processing you do in your DAW is saved. The only problem with this is the same as any other kind of high-quality file format: the increase in file size, in this case it is doubled over CD quality. But it is only necessary to save to this format during the production process up to the final export. When you export your finished track in the 32 bit format, this is where all those conversion utilities I mentioned before come in handy. Use the highest quality dithering algorithm you can get your hands on, such as Isotope RX or the Waves Ultramaximizer series (or failing that the default settings in your DAW's export menu) to convert your 32 bit master down to 16 or 24 bits.

If you have material at some oddball sample rate that is not CD quality, like 48K or 88.1K, well it's up to you if you want to convert it down to CD-Q or up to 96K or 192K, which is what most interfaces run at these days. Technically it's better to convert down, keep everything at a single rate, than it is to convert up and than back down. It's also less hard drive space, but it's again completely at your discretion what you want to do. Remember that a resampling algorithm can never "add" anything, it can only NOT take away what's already there. Also sometimes you have too many files, or too large of files, to resample easily. In this case just throw it into your DAW, use it's high-quality sample rate conversion setting, and focus on the music.

The most headache-inducing situation is where you have a session with audio at a variety of sample rates, as well as recorded material at a higher rate, again preferably the highest quality your interface supports. It helps to remember that a chain is only as strong as it's weakest link, the lowest quality element in your mix is what everything else hinges around. So to prevent headaches just run your session at the rate of the highest quality audio with the high-quality SRC mode in your DAW on. When your track is finished print all the audio at the session rate, mix it, and export it at 32 bit float. Than do a single sample rate conversion and dither with a mastering utility or r8 brain/SoX.  The latest version of Audacity is now using the sox resampling library, so yet another reason to check it out!

If you really need to conserve space, you can follow the same steps, but change the session rate to 44.1K at the mixing stage. That way your printed audio tracks will be much smaller in size, meaning less disk space used overall. You can also chuck the session onto an external drive when you're done, provided you've collected all the samples and saved (Your DAW should have some kind of setting for this) You should still record any recordings at the highest sample rate/bit depth your gear supports and keep them somewhere, as well as export at 32 bit float until the very last stage of the mixing/mastering process. If you follow these steps, you shouldn't have to think about audio quality, and you can focus on the content of your music. While it's nice to forget the technical aspects and craft your sounds, it's also important to be conscious of what you're doing. I mean, if changing some settings can improve the sound quality of your projects at the expense of a little disk space... Why not do it? disk space is only getting cheaper, why not be future-ready?

Well, that's just about everything I know about audio quality summed up in a practical manner. Hopefully I'll never have to revisit this topic until some new kind of technology comes out which renders everything I know obsolete. Until than...

Free music!

...and free Mumia! Seriously through, this is something I've been thinking about for a while, the idea of free and open source audio tools. One of the things I like about (digital) electronic music is that aside from the cost of a computer... It's basically free. You can download VST instruments and sample packs for free, and use a program such as Audacity to sequence and record your own sounds. This is one of the things I think is great about the digital revolution, it puts powerful tools into people's hands for little to no cost, compared to a studio full of physical gear.

But this brings up an interesting dilemma, touching not only the issue of piracy but going into the workings of  your computer and the digital audio workstation itself. It's all about the code, baby. When you start your computer and the operating system boots up, millions of lines of code are ran. Same as when you fire up your DAW, and those millions of lines of code need to inter-mesh with the hardware of your computer and audio equipment seamlessly for everything to make a sound. This takes alot of time, and it's very tedious.

So... Do you want to do this? You can if you like,  you can make your own software from scratch, but you probably won't be getting any music done for months or years. The easiest possible option is to pirate something, which takes minimal effort, and has no cost aside from your bandwidth (and an entire installation DVD, or a few, can start to eat it up quickly...) OR you can use free software, which can occasionally prove the old adage "you get what you pay for"-as in, it's definitely not as slick as commercial software. It's like eating home cooking versus something made at a restaurant. It has more charm, because it's more imperfect, and who could turn down a free meal?

One free meal I couldn't turn down recently was OpenMPT, which is also an open source program, meaning you can view the source code if you're nerdly enough to do so. It's pretty basic but if you're into early 90's computer culture or video game music you'll recognize some of sounds in the demo song that it comes with. I felt it was inspiring that people were into this oldschool style of sequencing program enough to spend this much time on a project of this scale. It made me think of how much time and effort went into making electronic music before the days of Ableton Live and Logic. On this topic, it's also inspiring in a way that a song originally posted as a demo on  the internet made it onto a charting pop song, even though how it got there was somewhat dubious:

The limitations that electronic musicians have put up with in the past lead to some great sounds being developed. At the same time as I now have this new program, with gigabytes of content, I also feel this sense of creative paralysis. Like a modern DAW overwhelms you with so many choices that when you open it up for the first time you start drooling out of the corner of your mouth while croaking "DAAAAWW..."-this is the epitome of the modern electronic music making experience. Breaking it down to a graph listing MIDI events & samples being played, like older programs did, is a great brain rinse. Especially when you realize you've spent hundreds on programs and plugins that you rarely use. The idea of free tools is appealing in that not only is it not costly, it is free of other restrictions such as copyright, and also guilt-free if they sit on your computer unused for a while.

So yeah, sorry about the rambling post. I would also like to apologize for my absence from this blog for the past week or two, I had a nasty cold on top of being hellishly busy at work. The combination meant I had to resort to all sorts of pharmaceutical garbage just to keep me online physically and mentally. I'm still recovering from it, and as you might guess I haven't got a chance to fire up Live 9 in a week or so. But now I'm back, installing the 6 DVD's of extra content, and I promise a review within the next few weeks.

Have a nice day, and keep your headphones on.

Placeholder Post: Live 9

I mentioned before I was going to do a review of Live 9, I have the program but I haven't got the time to dig into it. It comes with alot of new features, the least important of which is the off-shade grey interface. Looks slick:



The real "selling point" features are: audio to MIDI for both percussive and melodic material, improved EQ, a new compressor which is supposed to emulate the compressor unit from "a classic british console" Translated from marketing speak: it's a copy of the SSL bus compressor. Plus the Push controller, which promises to break the computer musician from the tyranny of the mouse and MIDI keyboard. This is something I'm not really into at the moment, but I have a friend who might be able to comment briefly.

All these are great and useful things, but what I'm most interested in is the addition of Max 4 Live to the Suite version of the program. Suite on it's own is great, although it was tad expensive up until version 8. But it's worth every penny, seeing as it's the flagship version of the program, fleshed out with enough sounds and instruments to stand on it's own with other DAWs. Adding Max support makes it a full package, in my opinion something was missing from Suite 8. Before Suite 9, adding the price of Max to the program would cost you ~$950 CAD in total, wheras now you get the whole deal for $750. Not bad.

I've been wanting to get my hands on Max ever since I read that Autechre were into it. Now that I have it, and it's integrated into my favorite DAW it's sortof overwhelming. Not only do I have a major "4 years in the making" update to wrap my head around, but the object-oriented aspect of Max as well. What I'd like the next phase of this blog to be is to log my progress learning Max, and sharing anything I learn about using Live 9. Take care, see you in the next few weeks... I can't promise when, the Matrix might take me under.

The Phantom 8 Bits

So after going through my phase of upsampling, I realized a couple things:

1) Whatever sample rate you're working at, your DAW is converting every audio file into a 32 bit float.
2) The highest resolution D/A converters can output is 24 bits, so you're losing 8 bits compared to what's in your computer. But 24 bit audio still has content down to -144db, meaning sounds far below the noise level of even the most silent of rooms. If you're using 16 bit audio with a 24 bit interface, than you're actually hearing the 8 "phantom" bits out of the 32 total. But...
3) You can't get something from nothing.

The last point sortof hit me when I realized just how much an audio file could change from the time it's recorded (or downloaded) to when it's in your finished track. You ideally want to be conscious of every change the file makes, and make sure that every sample rate conversion, or dither (bit depth conversion) is done at the highest possible quality. Yet even with a CD quality WAV, most files will eventually be converted into MP3, cutting out about ~70% of the audio data. This conversion is what is called a lossy process-Something is lost along the way, so you better keep a backup.

The same thing is true for a higher quality audio file you save at a lower rate. Once it's done you can't go back, even if you convert it back to the higher rate, it will be the same amount of content just "spread out" more, so to speak. This would be a good time to point out that your signal is only as high quality as the lowest quality link in your signal chain. The real mindbender is that even your monitoring setup will greatly effect the choices you make in terms of mixing, and especially mastering. Figuring out what's actually there in the file, is an odd and somewhat Platonic experience. Every way you have of listening to a recording effects it in some way, your CD player will add a small amount of noise, analog formats even more so. Even the best monitoring setup might give you a picture of the mix that completely disappears when you listen to it in your car.

It's technically true, but odd to say that you can't ever actually-hear-a 32 bit file, which is what most DAWs use these days. You can only hear a diluted version of what it is. So what's the real thing? Good question!

Keep It Simple, Smartass

Every once and a while it's a good idea to pull back and gain some context, see the forest for the trees. Most of what I've recommended with regards to keeping your audio at the highest quality possible, such as:  recording/mixing at a high sample rate & using certain re-sampling algorithms, can be disregarded without much consequence. I'm simply laying out some rules or theoretical ideals, that it would be nice to abide by. Whether or not you do is up to you. But as the old saying goes, it's a good idea to know the rules you're breaking before you break them.

Another thing to keep in mind, is that a vast majority of people listen to their audio in MP3 format. This is much lower than the CD quality standard I have been comparing even higher quality audio to. So logically it follows that even if you make a very good quality recording at CD quality, some of it's nuances will be lost when converting to ~320kbs MP3. So why bother? Because in the future, when storage space is even cheaper than it is now, and audio playback even higher quality, people will want better quality audio. Maybe not everyone, but the real music lovers will appreciate it.

With regards to MP3 listening, of course I listen to MP3's too. Having an iPod with thousands of tracks on it makes it too convenient not to. Ripping a CD to MP3 (or ideally AAC/MP4) takes minutes, so does downloading a digital copy from Bandcamp or another digital music marketplace. But mixing and mastering a track takes hours, days, weeks, months... and developing your overall musical style, chops, production techniques are all processes that take years. That all this work can all be condensed into a few minute download is incredible, and yet it makes even the most progressive music makers nervous.

Without getting into an entire other tangent on piracy, and the ethical issues that it implies, I will simply make an appeal to sound quality. MP3 is a very limited slice of the sonic spectrum, and even CD quality WAVs or FLACs aren't all there is to music. For certain styles a cassette tape would provide a more than adequate, and sometimes preferable listening experience. But the days of Walkmans and even Discmans are gone. It seems ironic that as the capacity to record music in ever higher quality becomes available to the average person, the quality of music being passed around has become lower. It is out of this paradox that my generation's taste in music is emerging from, and it is worth thinking about what a new standard for digital audio might look like.

That's all for now, keep your headphones on!

Sonic Dimensions

Someone mentioned that so far my blog has been more about audio than music. This is a very valid point, and I would say part of the reason I started this blog was to help myself get out of these loops of obsessing over sound quality, and getting down to what's really important-what the sounds are trying to convey. One of the ways I have been thinking about music lately is in dimensions. Time, frequency (or pitch) and volume seem to be the three dimensions of music as length, width, and height are in geometry. In traditional notation, time is represented in beats per minute, and the beats divided up into a time signature. Pitch is than represented by notes on the staff, tied to the tempo of the piece. Volume (or dynamics) at it's most basic is either piano or forte, but this could be an entire other discussion in itself.

That is in traditional musical notation, made for acoustic instruments. But when it comes to the main topic of this blog, digital audio, it becomes possible to control all of these aspects of sound. This can be thought of as either the greatest freedom, or an unimaginable prison. Time in digital audio is a function of the sample rate of the audio, measured in samples per second. Each sample is a tiny fragment of a waveform, which makes up all the various frequencies and their dynamic levels within a piece of music. This is a fascinating process that I'm still trying to wrap my head around.

A good starting point to conceptualize this process might be white noise, which is produced by generating random numbers at the given sample rate. So 44 100 random numbers between -32,768 to 32,767 for CD quality white noise. Everything is homogeneously random, no discernible pitch, no rhythmic/temporal characteristics, or dynamics. A constant tone such as a dial tone could be thought of as one dimensional, being a constant pitch with no tempo or dynamic change. Or a constant rhythm with no pitch and no dynamic variation. Adding another dimension might come in the form of varying one other aspect of the other two. It starts to get sticky here... but bear with me.

We've all heard someone play mary had a little lamb over a touch tone phone, right? Well compare this rudimentary musical performance to the dialtone, in comparison is has a variation in pitch that is recognizable as a melody, even though the tones do not exactly correspond with musical pitches. There is also a rhythmic component that accompanies the melody, making it recognizable, but there is no dynamic variation between the pitches. To be fully three dimensional music needs to have this dynamic component. A way of achieving this might be to sample the tones and than play them back in your DAW, thus breathing some life into this otherwise flat performance via the magic of MIDI.

What the fourth dimension of sound might be is something that I can only really guess at... Actually I know it, but it's a secret and if I told you your face would melt ala Raiders of the Lost Ark... It's also the secret ingredient in KFC's patented blend of herbs and spices. I'll leave this here as a clue:

Case study: Go [s]Plastic!

To take a refresher from all that math, I will now talk about the real world applications of sampling.

One of the things that surprised me when I looked into how some of my favorite albums were created, was that many of them weren't made on a computer at all! It has indeed been possible to make electronic music with just a tape machine and analog synthesizer since the early decades of the 20th century, no PC or Mac needed. This is something that AFX (Richard D. James) has explored in his Analord EP series, a project based around making acid house music using a completely analog signal path. This was how electronic music was made before technology made it more affordable to use computers than a studio full of hardware and a mixing desk. There was also a short period before computers took over when digital hardware was robust and abundant. Using only digital hardware modules means it is indeed possible to make music digitally without using a computer.

To illustrate this, I'll use an example of another Squarepusher album I enjoy, Go Plastic. Even though this album sounds super digital and computerized, it was actually made entirely with hardware. The gear list includes:

• Yamaha QY700 sequencer
• Akai S6000 sampler
• Boss DR 660 drum machine
• Yamaha TX81Z and FS1R synthesizers
• Eventide DSP4000 and Orville digital effects processors

(Thanks to wikipedia and "Rockin' On Magazine (Japan) 2004" for this info)

So I'll break it down and relate it to what I have explained thus far. The modern DAW is basically a self contained sequencer, meaning song arranger, as well as a sample player.  A DAW usually also includes some effects processing and synthesizers. But in this hardware setup, the Yamaha sequencer and Boss drum machine would be connected via MIDI to the Akai sampler and the two Yamaha synthesizers. The sequencer sends MIDI messages to the sampler and synths, and these modules generate sound which is output via their analog outputs, which would than be plugged into a mixer and amplified.

The sampler and synths were either connected to the Eventide effects processors directly, or via auxiliary feeds in the mixing console. Part of the "magic" of a setup like this would be the different combinations of effects & the variety of sounds one could get out of different routings. This is again something that can be emulated within a DAW, and each program has different ways of accomplishing this. To clarify, even though every one of these pieces of gear is digital, they have analog outputs thanks to digital to analog (or D/A) converters. Therefore they can be mixed and recorded to analog tape if one desires. Part of the sound of this record likely comes from the fact that every hardware unit has it's own set of D/A converters, which colour the sound on top of the colour the console is adding, which makes for a very colourful listen!

You could also feed the output of each instrument into an analog to digital (A/D) converter, or connect them via a digital output if they have it, and mix within software. This is called mixing "in the box"-in a computer as opposed to mixing "out of the box" on a console. The advantages and disadvantages, as well of the evolution of each, is something I will be covering later. But this is a good point to stop, take a rest, and ponder: If you can make music with digital instruments, than mix it and record it in an analog manner... or record some analog synths and live drums to tape, and than digitize it... What's the difference? What's going on here?

The difference lies in what happens when a sound is sampled, converted to digital, versus recorded to magnetic tape. These two processes are sortof like freezing versus drying food, they both preserve it but how? And what about the "loudness wars"-and what is digital clipping? Patience, all will be covered in time.

Just chill/GO sPLASTICk!

RoX ur SoX

As mentioned in my last post, it is indeed possible to convert between sample rates. But there is some loss of quality involved, so if you need to convert your audio between two rates you better have the best tools available if you want the best quality results. So after a colossal amount of sample rate nerdery, and close study of these graphs:

http://src.infinitewave.ca/

I have determined the best overall resampling algorithim is SoX 14.4 Very High Quality, intermediate phase. The binaries and windows .exe can be found here:

http://sourceforge.net/projects/sox/files/sox/

But be warned, it's command line! However worry not, I will provide some example commands, based on what I've figured out. The results are good enough to put up with the interface. Drop a file you want to convert into the directory sox.exe is in (or vice versa) than navigate there with the command prompt and enter:

C:\Users...> sox Amen44.1.wav Amen96K.wav rate -v -I 96000

This will tell SoX to create a new .wav file from Amen44.1.wav called Amen96K.wav, using very high quality noise rejection, and intermediate phase response, at the rate 96Khz. This will not touch the file's bit depth, it will be left at 16 bits, let your DAW worry about that. If you can't hang with the command line aspect there's also a standalone windows program called r8 brain which can do a more than adequate job, you can find it here.

That's all for tonight. Cheers!

LVL 0.1 [sampling]

So I've said that I would come back to the issue of analog to digital (A>D) conversion, and how this process can change the signal. This is the time for that! Speaking of which, time is another component of digitizing an analog signal-it needs to be sampled at a certain rate. This process is like taking a large number of "snapshots" of the audio, much like how there is a certain frame-rate for video. For CD quality this rate is 44 100 samples per second, or in other words the sample rate is 44.1 khz. Much like with video, it is also possible to use more frames per second to create a higher-resolution signal. Sample rates from 48khz, up to 192khz are also used to record audio, higher quality files being very helpful during the mixing and mastering process.

There is another aspect to a digital recording apart from the sample rate, called the bit depth. CD quality sound specifies a bit depth of 16 bits, giving the signal a potential of 65,536 possible dynamic levels. Once an analog signal has been converted to digital, the waveforms in that signal have been "frozen" in time at the sample rate, and in terms of dynamics at the bit depth that was selected while recording it. CD quality, although adequate during the advent of digital recording in the 1980's, is now no longer the defacto standard of digital audio. Ideally, one would want to record in 24 bit quality, as opposed to the 16 bits CD quality specifies. 24 bit audio gives a possible number of 16,777,216 dynamic levels, which is much better suited to reproducing the dynamics and subtleties of live instruments, especially drums and cymbals.

Along with increasing the bit depth to 24 bits, it is ideal to record at a 96khz sample rate instead of 44.1khz. Although many find this step unnecessary, pointing to the Nyquist–Shannon sampling theorem which states that as long as a sampling rate is above twice the highest frequency being sampled, you're good to go. This is only partially true. The simple reason why 44.1khz was chosen as a sample rate was that it humans can only hear from 20hz-20khz. The highest frequency CD quality audio can sample is the sample rate divided by two:

44.1 khz/2 = 22.05khz

22.05 khz being well above what we can hear, which is ideal as the top khz or two will be lost in the digital conversion and filtering process. This makes CD quality audio "good enough" for most purposes. But why do professionals record at rates up to 192khz, even though nobody can hear up to 96khz? The Nyquist frequency is only half the story, when one samples at a higher rate, one also divides up the signal into smaller and smaller portions. This means that the signal can be captured more accurately, which is essential in the digital recording process.

Even if one chooses to record at the CD sample rate, it is ideal to use 24 bit audio, though your files will end up being larger. Since I've already got into some math, I will show you the formula which lets you calculate the size of files recorded at certian samples rates and bit depths. (From the Wikipedia article)

"Bit rate = (sampling rate) × (bit depth) × (number of channels)"

"The eventual file size of an audio recording can also be calculated using a similar formula:

File Size (bits) = (sampling rate) × (bit depth) × (number of channels) × (seconds)"

To fill one of these in for an hour of stereo CD quality audio would go like this:

44100 x 16 x 2 x 3600 = 5080320000 bits

Divided by 8, since there are 8 bits in a byte,  gives us 635040000 bytes, which is 605.6 MB

To up the quality to 24 bits like has been suggested would give a filesize of 908.4 MB for an hour of stereo audio at the same sample rate of 44.1 khz. Increasing the sample rate to 96 khz would yield a filesize of 1977.5 MB or 1.93 GB. Compared to CD quality audio this is a 2.45 increase in filesize. It's up to you what you decide to fill your hard drive with, if the sound is really that much better to justify the increase in size of your projects. Again, this only applies to recordings you make yourself, anything that is already recorded at it's chosen sample rate and bit depth is "frozen" there, although it is certainly possible to convert between two rates, with some negligible loss in quality.

The Basickz

The software equivalent of an analog console is known as a Digital Audio Workstation, or DAW as I will commonly be referring to in this blog. The main difference is that an analog console is made of wires, capacitors, and transistors, it is a physical piece of equipment. A DAW is a piece of software that is designed to emulate many of functions of an analog console in digital form. To get audio into your computer you first need to convert it to a digital signal with an analog to digital converter, or A/D converter. I will go into the specifics of this process later, but all you need to know at the moment is that this is where digital and analog recording part ways. This step is not necessary if you are recording completely analog. Both analog and digital recording involve recording many tracks of audio, and than editing and splicing them together. The difference lies in how this is done, and what happens to the signal along the way.

To recap, this is an analog console: 

This is a DAW:

There are obvious differences between the two, but also some similarities. One is physical and one is not, yet they both have faders and means for editing. The bigger difference is in the cost, which is something I don't even want to think about. Both an analog console, effects, and tape machine, as well as a Pro Tools HD rig cost many many times what my current setup does.

For the record the DAW I use most often is Ableton Suite 8, and I am looking forward to version 9 which is coming out on March 5th. Expect a full feature once I get my hands on it. Ableton has been around since 2001 and focuses on sampling and looping audio as well as arranging MIDI rather than recording audio, although it is fully capable in that respect. Another program I use, or try to use, is Renoise, which is a DAW not based around emulating a recording console at all, but comes from the heritage of tracker programs which goes back to the early 90's.


Here are some links if you want to check these out:

http://www.ableton.com/
http://www.renoise.com/

Hello [Goodbye] World!

Alright so I've decided to create this blog in order to dispel rumors and myths about digital audio. I would also like to provide a little counterpoint to the many voices out there singing the praises of analog audio. There's enough of that both on the internet and off. For fairness' sake I do happen to own a tape machine and some small analog mixers, as well as some basic digital recording gear. None of the gear I have is considered professional quality, and I do not consider myself a professional audio engineer or producer. The perspective this blog will be written from is that of an amateur musician who is interested in the production aspects of music, and would like to "cut through the hype" and get good sounding mixes as quickly, easily, and cheaply as possible.

One of the main things that sparked my interest in this subject was watching Dave Grohl's documentary Sound City about the legendary Sound City recording studios in San Fernando Valley. One of the key arguments in this documentary was that the studio's Neve 8028 analog console was a huge contributor to the overall production process. This is something I can understand, as technology has been a crucial part of the evolution of music over the past several decades, especially multitrack recording technology. However, I felt there were some elitist undertones to certain parts of the documentary. I can understand preferring the sound of analog electronics and recording, especially if you grew up listening to music that was produced this way. But some of the musicians and engineers were saying that the lower cost of digital recording gear as well as the internet, have introduced artists into the mainstream that "should not be heard"-Says who?

In my opinion, anything that takes away this "wall" between what the public hears, and what musicians and artists are actually producing is a good thing. This is a separate, somewhat more political and economic point than the technical issues this blog will deal with in the future. But I feel it is important to say, as I would probably not have heard of many of the artists I enjoy without the internet. I also really enjoy hearing what artists can do with the tools modern digital recording technology offers. What I intend to do with this blog is create a kind of "safe space" where I can riff about the things that matter to me. I've noticed few places on the internet cater to the kind of music I care most about, so I will do my best to stimulate the minds of anyone who comes across this blog, just as my favorite musicians have done the same to me.

Here is a great example of a perfect marriage between digital and analog technology. Tom Jenkinson aka Squarepusher, together with his Akai S-950 sampler and bass guitar, creates an utterly alien and beautiful sound. When I first heard this it blew my expectations of both jazz and electronic music out the window. This album show it is certainly possible to have both sequenced, pre-programmed material, as well as live analog recordings get along in the same sonic space. Why? Why not!

So it is with this spirit that I say... Hello [Goodbye] World!