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.