Well… it’s been quite awhile since I’ve been active on TheStudioFiles.com.  It was a busy year – full of world travels and tons of audio work.  Luckily, my schedule will now allow me to return to TheStudioFiles.com and resume articles, videos, Q&A’s, and general tech support.

In other news, Avid has gone through some major changes and upgrades while I’ve been gone.  Most notably – we have the release of Pro Tools 9 along with a new line of interfaces for all lines of Pro Tools.  As you’ll see in upcoming articles, I’ve actually just upgraded to HD Native on an Avid 16×16 analog HD I/O – and I love it.

So – sorry for the hiatus – but I’m back!  I look forward to another great year with The Studio Files.  If there’s anything in particular you’d like to hear about or see in upcoming articles/videos, please let me know in the comments or via email.

Connor

TheStudioFiles.com

Its not much, but DigiTechSupport (after an 80+ page thread at the DUC) posted:

“You should, however, start to see some fruits of this labor over the next few months, but from a slightly different angle than you may suspect.”

Hmm… now we can all prod and speculate on what that angle may be… 

On a side note – I’d like to thank Digidesign for at least having presence in the forums and answering what questions they can.

C

The Studio Files

Apple announced Tuesday an update to the media suite iLife during their Macworld 2009 keynote presentation.  Here is an early list of some of the new features announced.

iPhoto ’09 – Support for GPS geo-tagging (with Google Maps), face recognition, built in uploading to Facebook and Flickr.  The face recoginition will actually remember faces that you’ve tagged in Facebook and tag photos automatically!

iMovie ’09 – Advanced and easier to use editing, true drag and drop controls, image stabilization, more video effects and Indiana Jones style animated travel maps!  Apple actually acknowledged the uproar over iMovie ’08 (the majority of users considered it a downgrade from iMovie ’06) by stating “it was missing some features.”

GarageBand ’09 – Built in video lessons for piano and guitar and downloadable “Artist Lessons” for $4.99 each where the artist teaches you how to play their song (for example Sting will show you how to play guitar on “Every Breath You Take”).

Hopefully we’ll see some other additions to GarageBand ’09 such as more software instruments and improvements to the interface.  Stay tuned to the Studio Files for information on GarageBand ’09 as its released.

-PJ

The rumor mill has been swirling for months over at www.appleinsider.com with reports that the Mac Mini will get a much needed revamp, along with new iMacs featuring Intel Quad Processors.  It has also been reported that iWork is due for an update and I personally think that iLife will see an update as well.  Perhaps in light of the new Pro Tools release, Apple will make an announcement regarding Logic and Logic Express too! Hopefully, since this is Apple’s last year at Macworld, they’ll go out with a bang.  We’ll keep you posted.

Have any expectations or predictions for Macworld 2009? Tell us about it in the comments!

-PJ

www.icompositions.com

This is definitely one of my new favorite websites.  I stumbled upon it one day whilst googling “place to share garageband songs” and found just that.  The main music area is very active with about 50-100 songs uploaded a day or you can sort it out by your genre of choice.  Add in the slick comments and ratings system and you have an excellent site where you can get instant feedback on your tunes or just listen to other users’ original music.

The site was redesigned in October giving it a new look and much faster upload speeds for songs.  The upload form, which is one of the big reasons I like the site so much, lets you upload MP3 and/or AAC, title songs, write a description, select a genre, select the program it was created with and other miscellaneous details on how to handle comments and song rights.

Creating an account gives you access to the Backstage area where you can manage your profile, song library and favorite songs and artists lists.  To give you a better idea of what the profile page looks like, here is mine:

http://www.icompositions.com/artists/pjATthestudiofiles

EDIT 12/30:  Heres a couple artist profiles I would recommend checking out also.  I’m a big fan of these guys and even collaborated with Waleed on track (more to come!) 

http://www.icompositions.com/artists/Waleed

http://www.icompositions.com/artists/FEAHM

In addition to the tunage, iComp also has an active forum and live chat.  All this makes icompositions.com a winner in my book and I recommend that any audio enthusiast or hobbyist give it a whirl.

Have a website that you would like featured on the Studio Files?  Know of a site that you’re just dying to tell people about?  Send us an email or drop us a comment below!

-PJ

Knock ‘em dead C-dog(s)! Wish I could be there.

Damon
The Studio Files

Also remember to keep your browser on thestudiofiles.com after December 10th.  Pro Tools 8 ships on the 10th, and as soon as we get our copies, expect articles and videos covering all of the new features.

Working in an industry where it’s your job to listen to music all day, knowing the safe limits of our hearing system is really important.  You only get one set of ears, they need to last!

The ears can safely listen to a WIDE dynamic range of sounds.  However, the duration of this “safe” window decreases rapidly as sources increase in volume.  Many of us in the audio industry will be listening for the better part of a whole workday (say 8 hours at least), so its good to know how loud we can listen without damage.  The problem is that everything sounds better when its louder.

OSHA (the Occupational and Safety Health Administration) is a federal organization dedicated to safety in the workplace.  OSHA has generated a list of dB ratings with associated time durations which will not damage the hearing system.  Unfortunately, these figures are sometimes disputed.  OSHA first outlined that it was safe to listen to 90 dB sound for 8 hours per day.  However, many audio professionals think that the number should be 85 dB.

In my opinion, its better to operate on the safe side (right?), you won’t miss those extra 5 dB.  There is some debate on how long to listen to music at certain SPLs.  Taking in account factors such as the natural compression of the middle ear, the duration of breaks, spectral content of material, how our ears deal with loud sounds (vs. soft sounds), etc. can factor into these numbers (as well as the weighting of the SPL meter).  Here are some basic guidelines for listening levels (each of these being “per a 24 hour period”):

• 8 hours at 85 dB
• 4 hours at 93 dB
• 2 hours at 100 dB
• 1 hour at 105 dB
• .5 hour at 110 dB
• 15 minutes at 115
• Anything higher, get out immediately!

Also remember, these are over a 24 hour period.  Once you listen to 15 minutes of ~115dB music, you need to shut your ears off from the world for 23 hours and 45 minutes to ensure no damage occurs.

Another good thing to keep in mind is that relationships between decibel levels are logarithmic (not linear).  So if you jump from 50 dB to 110 dB, thats not 2.2 times louder, its 1,000,000 times louder (ONE MILLION!).  Crazy huh?

The moral of the story: buy an SPL meter and use it!

SPL meters are inexpensive, and can save your ears.  Keep one by your mixing desk and check it occassionally throughout the day.  Its also good to calibrate your listening system so that you know where certain levels are located on the knob (relatively close obivously).  Setting your “normal volume” to around 80dB will ensure that you are safe for the whole day.

Anyone working in a studio knows that sometimes during the day, its time to listen to the song really loud once through (not only to pump the band up, but also to hear what the song sounds like at that volume).  This is fine!  Just make sure it doesn’t drift above 100dB too much, and that you substitute some sub-80dB listening in parts of the day to make up for it.

More on SPL meters and decibel ratings in the future.  Any quesitons? Post them here.

C

The Studio Files

Being a prepared and efficient musician can save money, increase happiness (for everyone!), and help the album prosper musically.  Being an unprepared and inefficient musician can cost money, decrease happiness (for everyone!), and put a negative vibe on the album.

What to bring to a session

• Yourself, on time (=early).  Arriving early will give you time to meet the engineer/producer, setup, get your instrument warm, and make everyone more comfortable.  Tardiness=ouch!
• A chart of the song.  Unless you know the song forwards and backwards without fault, its nice to have a cheat sheet, even if its just the major chords and section changes.  The less you know the song, the more detailed your chart should be.  Figuring out chords/changes when you should be recording can put a damper on the session.  I can’t emphasize enough that you will be in high demand as a musician if you consistently come to the studio already knowing the music well.
• If you are a guitarist (or bassist, etc) be sure to bring extra strings, picks, cables, a strap, a guitar stand(!), and a tuner.  In my experience, the most often forgot items are tuners and guitar stands.  Tuners are obviously essential, be sure to use them frequently!  Guitar stands, while not as obvious, are also a good idea.  Propping your axe up against a chair or the console can only lead to disaster in a studio full of people, cables, (i.e. things that can knock your guitar to the floor)
• If you are a drummer, bring extra sticks (of different weights if possible), extra damping (O-rings, moon gel, etc), and maybe even some WD40 (for that squeaky kick pedal, etc).
• If you are a vocalist, make sure your throat isn’t torn up from screaming the night before.  Be sure to get there early to do some vocal warmups.  If you know its going to be a long session with lots of singing, bringing extra water, tea, olive oil (a quick shot can really help the throat, as gross as it might seem) will keep your throat ready to rock.  Also bring copies of your lyrics.  This fulfills two purposes.  First, you won’t forget your words.  Second, the producer can follow the lyrics as you track to help decide any musical suggestions (as well as to help pick takes).

Those^^ are just some generalities, but they are at least a good start.

The most important concept here is to be prepared to play the music.  Know the changes and follow the groove.

Some things to not do:

• Whether you are between takes or someone else is doing overdubs, don’t “noodle” around on your instrument just to make noise.  Often this can cover up any communication between the engineer/producer and artist, as well as irritate the heck out of anyone in the control room.  If you need to practice a lick a few times before the take (which of course is more than acceptable), just be sure to tell the engineer what you are doing.
• Once you are setup, try not to change things on your instrument without alerting the engineer first.  Changing the gain on your guitar’s distortion pedal may mess up the engineer’s signal chain the other room, etc…

That’s it for now, more on this to come

C

The Studio Files

This article will make more sense after reading the rest of this series (Take 1, Take 2, Take 3, Take 4, Take 5, and VBR)

The following audio files were made by converting a CD quality (44.1/16) version of Steely Dan’s Hey Nineteen into compressed formats (MP3, OGG, and FLAC… still tracking down my AAC file…).

Once converted, each file was summed to mono, polarity flipped, and summed with the original WAV file.

Now – this is NOT a representation of what “MP3 takes out”.  Its not that simple.  These compression codecs also introduce new distortion, restructure the bit depth, time shift peaks, raise/lower noise floors, etc… Therefore, this is not all that much of a scientific comparison.  So, then why are we doing it?

In essence, tests like these give you an idea of what the failure mode and distortion (aka, the “yuck”) within the compressed formats sounds like.  Also, you can at least get a good idea of how the compression codec is dealing with the high end of the spectrum.  Generally, the more high-end you hear in these examples, the more the codec cut out of the original file when compression down.  There are 2 specific things to notice in these.

1) Compare the mp3 to the Vorbis.  Notice how the Vorbis sounds more analog-like?  (Much more like natural white-noise-esque analog noise, rather than the harder, digital sounding distortion of mp3)  If you refer back to the article on Vorbis, you’ll remember that Vorbis alters the level of the noise floor to mask nasty-sounding distortion.

2) The FLAC summed file may not be what you expect.  Remember… its lossless (!hint)

(haha, its funny to note that the following files are actually .mp3… This is a great demonstration of why we need these formats… WAV files are too big!)

256 KBPS MP3 flipped and summed with original

256 KBPS Ogg Vorbis flipped and summed with original

FLAC flipped and summed with original

Questions? Disagreements?  Post it in the comments.

C

The Studio Files

aka FLAC (Free Lossless Audio Codec)

Keyword there being: Lossless!  No data is removed from the audio.  Let’s dive in…

Again, if you haven’t read Take 1, Take 2, Take 3, Take 4, and VBR, you may want to…

So far, all the compression codecs we’ve looked at are lossy, meaning they actually remove data from the file (and therefore, you could never accurately convert back from the uncompressed file).  The other encoders work on the principle of masking and psychoacoustic models, looking to “take out” ot de-emphasize the things humans won’t hear.

FLAC doesn’t.  And while similar in some ways, this is a whole different breed of encoder.  Plus (like Ogg Vorbis, its open source = FREE)

For starters, FLAC (like the lossy encoders) breaks the signal into smaller pieces.  The length of each piece is based on a whole slew of different factors, including dynamic level, spectral content, transients, etc… Now, this is the point where the lossy encoders would start axing (leaving out) things.

FLAC’s goal is to formulate its best prediction of how the signal is acting in the frame.  Once it has the best mathematical prediction it can come up with, it compares the prediction to the original, and encodes any differences (called the “residual”).  Check out this picture – its a verrrrrrrrry basic representation of what is going on here:

Now, remember that digital audio is a very finite thing.  At every given sample point, there are only so many different positions (within the bit depth) that the sample can be at.  This is essential to understanding how FLAC could be lossless.  If we were dealing with a signal that did not have a finite number of points (aka, analog), this would not work.  It could be close, but not completely lossless.  By having a mathematical representation of a frame (plus the residual encoding), FLAC can shrink the file size down usually around 30-50% (but more in some circumstances).

When I was first delving into FLAC, I thought “well… how can a compression format that is lossless have different levels of compression???  It must not be lossless…”

It has to do with CPU cycles.  When compressing FLAC as much as possible (say you are going for 50% size reduction), the computer will spend much longer trying to predict what the waveform is doing.  By getting a more accurate prediction up front, there will be much less residual data to encode.  So, the trade-off for a smaller file size is a longer encoding time (and be aware that I have not been able to find a FLAC encoder that can access more than one core of a processor).

That being said, we are only talking a few seconds here.  Where an mp3 may encode in 5 seconds, a fairly high compression FLAC might take 25 seconds.  If you read Part 4, you will have read the introduction to why open source compression formats aren’t amazingly popular right now…  If you google FLAC, you can find a slew of information on encoders, decoders, players, plug-ins… etc.  With newer iPods, I’ve heard there’s even a mod you can do to get FLACs to play (although not on the old ones, the processor requirement is too high) – careful though, I’d bet this is not within any limitations of the warranty or user agreement…

So what the heck is FLAC useful for?  My first thought would be for archiving mix/mastering files.  With 50% reductions on the files, all of a sudden you have to buy half as many backup drives.  It’s also useful for sending high quality audio to clients over the internet with less bandwidth (assuming they also know how to procure a FLAC player)

Hopefully we will see FLAC grow in popularity in the near future!  More to come soon on these compression formats…

C

The Studio Files

]]> https://thestudiofiles.com/?feed=rss2&p=387 0 https://thestudiofiles.com/?p=376 https://thestudiofiles.com/?p=376#comments Wed, 19 Nov 2008 02:21:42 +0000 http://thestudiofiles.com/?p=376 Ogg Vorbis?

Definitely wins the award for the weirdest name among digital audio compression formats.  However, Ogg Vorbis is arguably one of the best sounding and easiest to access of the bunch (it’s open source, aka FREE)

Again, this article assumes you’ve read Take 1, Take 2, Take 3, and hopefully also VBR.

“Vorbis” is actually the audio compression codec.  It was developed as an open source (lossy) compression codec as a substitute for mp3.  “Ogg” is simply the file container in which the Vorbis is held, thus “Ogg Vorbis”.  (Its worthy to note that you can hold lots of other things in Ogg containers, including FLAC).

So what’s different about Vorbis?  For one, its open source.  That means there are no licensing costs associated with using it.  Unfortunately, since people aren’t making millions of dollars off it, the effort to significantly popularize (or even standardize) the format is a hard sell.  MP3 and AAC are very much market-driven formats (i.e. iTunes uses AAC).  Since companies and individuals have incentive to make large amounts of money off of their codecs, its no wonder that MP3 and AAC are overwhelmingly more widely used than Vorbis.  That being said, it is good to know that Vorbis sounds really good at equivalent bit rates, AND its free.  (So although you’d be in the minority, if you are using Ogg Vorbis, you have something that sounds better* and costs nothing to license).

It’s also worth noting that if Vorbis were licensed and AAC open source, we’d likely be saying the same sort of things the other way ’round… this is really just an open discussion.

So, other than being free, what’s good to know about how Ogg Vorbis works?

Vorbis, like AAC, uses an MDCT to begin to break the digital audio into small pieces, and put them through the frequency-analysis process.  Vorbis uses its own psychoacoustic model (which theoretically is better than its predecessors).  Of course, the implication here is that if the psychoacoustic model is “better” than AAC and MP3, it will be getting better sounding results at the same bit rates.

Vorbis is different too in that it is always variable bit rate. (Click here to revisit the article on VBR encoding) While this will yield more pleasing sonic results for Vorbis, it can make it more difficult to compare MP3 and AAC encoders to Vorbis objectively (as some are not VBR).  Now for the big difference…. drum roll please….

Vorbis is different in its “failure mode”.  Here, I am using “failure” to describe what happens with the encoder doesn’t have enough available bits to get the good audio encoded above the distortion.  Remember, the encoder is inevitably going to have to deal with noise from things like quantization distortion, and ideally, all of that will be below the masking threshold (aka, we won’t hear the bad stuff because the good stuff is loud and covers it up).  When, let’s say, and MP3 encoder fails, the distortion gets loud, you may hear metallic swirlies (also called birdies, think “youtube audio”), other non harmonic distortion…

But! When Ogg Vorbis fails, something entirely different can happen.  When the audio is initially going through the filterbank in Vorbis, the encoder separates the noise floor from the audio.  When the encoder would fail, the noise floor is raised.  This gives a much softer (almost analog-noise-like) distortion, rather than birdies and harsh distortions.  So, between an improved psychoacoustic model, the inherent VBR, and the nicer-sounding failure mode, Ogg Vorbis seems like a great choice.

Again, problems with Ogg Vorbis mainly stem from the fact that it is not a market driven format.  There are many plug-ins and stand alone players that will read and encode/decode Ogg Vorbis files, but they are not nearly as numerous as those of MP3 and AAC files.  Most portable music players won’t directly play Vorbis files without some software tweaking.

FLAC is next!

Also coming soon are flipped polarity examples of these encoders with the original files (so you can hear the failure modes, and other “yuck” left over/added from compression conversion)

C

The Studio Files

…which, although commonly referred to as “Apple Audio Codec”, this actually stands for Advanced Audio Coding.

(This article assumes you have read Everything About Compression: Take 2)

Although originally coming out of MPEG-2 specification, AAC is known as Part 3 of MPEG-4. AAC was designed as a newer, “better” sounding alternative to MP3. Generally, the AAC encoder follows the same steps as the MP3 encoder, but let’s take a look at what makes the two formats so different.

First off, the filterbank (used to “chop up” the original audio file and determine the frequency content) in the AAC encoder is not a hybrid filter like the mp3 (which uses both an FFT and and an MDCT). The AAC encoder just uses an MDCT (the most important thing to note here is that an MDCT will typically overlap its bands of analysis, thus reducing any errors that might come at the “edge” points between blocks). AAC’s MDCT also has greater resolution than MP3′s (1024 vs. 576). AAC can also switch its block size (the length of the audio pieces it is analyzing).

The AAC also has a superior psychoacoustic model. The model was developed later (and likely spun from the MP3 original research). Presumably, the encoder with a more accurate perceptual model will be able to achieve better quality at lower bit-rates (kbps). The perceptual model is arguably the most important element in an encoder, so AAC’s improved psychoacoustic model is the most prominent reason why AAC sounds better at identical bit rates to mp3. (Note: “better” is obviously very subjective an open to debate. This is speaking rather theoretically)

Finally, AAC also has some additional tools which can help with efficiency, including Temporal Noise Shaping (TNS) and a Prediction stage. The inner processes of TNS are a little beyond the scope of this entry… but basically its a type of noise shaping and prediction that was originally implemented to help improve the sound of speech at lower bit rates. It helps with the efficiency and quality of the encoding. The prediction stage is not always used, as it is most useful in signals that are easily predictable (sine tones or tone-like soures).

Phew… enough of the technical jumble…

Oh, I should also mention that most AAC encoders will have the option to encode with variable bit-rate (which I went into in the VBR Explained Article).

So that’s AAC…

C

The Studio Files

This is a tangent from the compression articles. I thought I should explain VBR.

Normally, when encoding a compressed audio files (let’s say, an MP3), you need to specify a bit rate in KBPS (kilobits per second). This tells the encoder how many bits it has to work with over the given time period. (See Everything About Compression: Take 2 for info on what the encoder is doing)

But, what if a REALLY complicated second of audio is followed by a really simple second (say a second of metal mathcore followed by a 1k sine wave). In this case, the encoder would struggle to fit all the metal into the allocated bits, but would probably have a lot of wiggle room with the sine wave encoding.

Well, a VBR encoder would realize this, and vary the bit rate for that section. It would allocate more bits to the second of complex signal (the mathcore), leaving less (but more than sufficient) bits available for the sine wave encoding.

To relate it to real life… You ate a sandwich (on wheat bread) that had 300 calories. Now, you have two things on your “to do” list:

1) Stand up

2) Do 9007 jumping jacks

If you were not a Variable Bit Rate encoded person, you would allocate 150 calories for standing up and 150 calories for doing 9007 jumping jacks. What’s going to happen? Standing up is no problem, but sometime after you start jumping jacks, you are going to fail. When the compression encoder “fails”, it does things like not giving enough bits to the audio signal to raise it above the (quantization-induced) noise floor, thus audible distortion.

If you were a VBR encoded person, you could allocate 1 calorie to standing up, and 299 to jumping jacks, thus giving yourself almost twice the chance of finishing them.

A rather ridiculous example… but hopefully it gets the point across.

C

The Studio Files

- More in the Compression series (I’ll hit AAC, Vorbis, FLAC, maybe others).

- Basic human hearing physiology (…ok… I know… its mega-boring… But, I think its essential that producers and engineers know at least the basics of this stuff)

- Studio Etiquette!  I’ve been conjuring up some topics for awhile, I think this is a good one.  While a lot of it may seem super obvious, I think its good to “get it out there”.  1st topic in this thread will likely be “TalkBack Mic Etiquette”

Others worth considering – basic electronic components, impulse response/convo reverb explained, ad infinitum

If anyone has something they really want to see, let me know.

C