mp3 or AAC?

I’m surprised that so many music sites insist on using mp3 format for their files despite the generally poor quality – even Soundcloud uses only 128kbs mp3 – when AAC is a patently superior file format at that same file size/bandwidth. Most noticeable is the retention of transients (eg drum beats) and much better top-end response. AAC sounds almost like CD-quality.waveform2

Key points

  • mp3 is generally MPEG 1 (layer 3). They did extend it slightly in MPEG 2 to add some lower sample rates and some different channel formats.
  • AAC (Advanced Audio Coding) is MPEG 2 and also in MPEG 4.
  • mp3 only uses 576 blocks to encode audio.
  • AAC uses 960 or 1024 blocks to encode audio.

Note: MPEG stand for Motion Picture Experts Group – the standard-setting body for this sort of stuff.

There’s a huge difference in quality between the formats – mainly because they tweaked the hell out of the encoding algorithm after the original mp3, so 128kbps AAC sounds almost like CD-quality, whereas mp3 128kbps sounds phasey and dull.

AAC is also what’s generally used nowadays for radio broadcast – they call it MP2 though, to confuse the production people.

So what about MPEG 3 and 4 then?

MPEG 3 only had a couple of tweaks to add to the standard, so rather than having a completely new standard, they just updated good ol’ MPEG 2 instead. They added things like more channels for surround-sound files and the like.

MPEG 4 was mainly about metadata (information embedded in the files) and even more channels – basically bundling a variety of video and audio formats (and other stuff like subtitles) into a handy container called “MP4”.

If there’s only audio inside it, it gets called M4A. Though some M4A files can also contain Apple Lossless format at master quality, but at a larger file size. Lossless formats are more like zip files – they’re compressed down to smaller size (not as small as mp3 or AAC) but can be expanded again without any loss.

What’s this bit-rate thing?

Since most of these lossy formats were designed to be streamed – either on the internet or off an optical disc, they’re measured in how many bits per second they need.

Is that the same as sample-rate?

No – mp3s and AAC files still retain the original sample-rate and bit-depth as the original file. At lower bitrates, obviously more information needs to be trimmed out of the audio to make it smaller – hence the quality loss.

So why do we still use mp3 rather than AAC?

Well, if you’re using Apple products, you’re likely using AAC way more than you know. Otherwise, mp3 has just been around longer and more software can be guaranteed to play mp3s than AAC (although it’s pretty rare that something can’t play both now). It’s like the old VHS vs Betamax video tape format thing all over again – the lowest common denominator (and cheapest format) usually wins against quality.

So what should you use?

If there’s a choice – go with AAC. If not, go with mp3. Simple.

(Or a lossless format if possible!)

Tighten up your mix by using track-offsets

One of the cool things you can do so easily in a DAW (Digital Audio Workstation) is slip entire tracks, or actually the regions within it, to the left (earlier in time) or to the right (later in time).

What’s so cool about this?

There’s a couple of things you can do with it.

1. Correcting Microphone Delays

One fairly obvious example is tightening up drum kits. It’s pretty common to use room mics when recording a drum kit. Let’s say your room mics are 3m (9.8feet) from the snare mic on the kit. That’s about an extra 8 milliseconds, or (@44.1kHz sample rate) 353 samples.

If you slid your room mic regions 8ms/353 samples to the left, the recorded signal would coincide perfectly with the snare mic – ie there would be no delay between the mics.* Note that you’ll need to have all your tracks starting a little bit later than 0′:00″ or Bar 1 so you have some “left” to go to.

You don’t have to pull the room mics all the way back though – you might just want to pull them a bit closer in time to tighten up the room sound.

2. Creating a Pocket/Helping the groove

This is where you can subtly shift the timing of regions to help the overall groove of the track.

For example, many bass players get a little excited when recording, and can play a little ahead of, or right on top of the drum beat. Although it’s still “in time”, sometimes delaying the bass very slightly can make it “groove” more with the drums.

In this case you would incrementally slip the bass region/s to the right until the groove feels better. The kick drum often masks the attack of the bass to a certain extent, so it can also clean up the kick/bass combination.

It can help to think of being “in-time” as a window rather than a vertical line, and you can be at one edge or the other of that window and still be in time, but get some huge changes in “feel”.

How to do it

There’s a couple of methods.

Shifting the actual region itself using “nudge”. In Logic or Pro Tools you can incrementally bump something in either direction by using the nudge keys. You can set the nudge value to pretty much anything -samples, beats etc. The dangers of this method are if you give the files to someone else to mix they may not realise you’ve moved track regions away from the same start position. Keep good notes!


Region Delay in Logic Pro

In the region inspector (under “more”) there is a parameter for Delay. You can set it earlier (- values) or later (+ values). The handy thing about this way of doing it is that you can split each track into different regions for chorus/verse etc, which can each have a different delay value.

Using delays. This is the old-school way of doing it. Insert a delay plug-in (with the same delay value eg 500ms) on every track. If you want to make a track play earlier – lower the delay for that track. If you want it later – make the delay longer. The beauty of this method is that it automatically keeps a record of what you’ve done by saving the plugin settings.

*Phase problems. Watch out for this. Once you get down to a few samples difference between different microphones on the same instrument (eg the drum kit), you’re potentially going to get phase issues. Sometimes even a few samples can make a difference. The wavelength of around 2cm (almost an inch) is 13.5kHz, so that means moving the microphone “virtually” even that much can make a big difference. This is where phase “rotation” plugins can be handy – such as UAD’s Little Labs IBP Phase Alignment Tool.