For information, tips, and examples of use of this very handy image editing tool please visit: Pixelmator graphics tool zone.

A printer-friendly 300dpi version of the Webel IT logo for business cards was composed in Pixelmator, with very slight bevel effect, and slightly more pastel coloured letters than in the original web version:

ImageMagick (convert)

From ImageMagick®:

ImageMagick® is a software suite to create, edit, compose, or convert bitmap images. It can read and write images in a variety of formats (over 100) including DPX, EXR, GIF, JPEG, JPEG-2000, PDF, PNG, Postscript, SVG, and TIFF. Use ImageMagick to resize, flip, mirror, rotate, distort, shear and transform images, adjust image colors, apply various special effects, or draw text, lines, polygons, ellipses and Bézier curves.

The functionality of ImageMagick is typically utilized from the command line or you can use the features from programs written in your favorite language. Choose from these interfaces: G2F (Ada), MagickCore (C), MagickWand (C), ChMagick (Ch), ImageMagickObject (COM+), Magick++ (C++), JMagick (Java), L-Magick (Lisp), Lua, NMagick (Neko/haXe), Magick.NET (.NET), PascalMagick (Pascal), PerlMagick (Perl), MagickWand for PHP (PHP), IMagick (PHP), PythonMagick (Python), RMagick (Ruby), or TclMagick (Tcl/TK). With a language interface, use ImageMagick to modify or create images dynamically and automagically.

Although the user interfaces for ImageMagick are rather clunky compared with other image editing tools, there are time when it is very useful to have its command line capabilities to manipulate an image or sets of images - such as command-line driven batch processing - and this is where ImageMagick really shines. It can sometimes offer a quick command-line rescue for a task that another tool can't handle.

It is sometimes simply known as convert, because that's the best known of the command line commandsit offers.

It is also handy to have programming language interfaces to it, so that, for example, images can be manipulated on-the-fly on a web server in the PHP language.

The easiest way to install it on Mac OS X is as a MacPort:

sudo port install ImageMagick

GraphicConverter tool for Mac

GraphicConverter is a must have for anybody working with graphics on Mac OS X, no matter what other tools you use for graphics editing, drawing, or image manipulation. It is a "Swiss Army Knife" for graphics manipulation and conversion, as well as basic editing, but it's strength is in conversion.

It can, for example, apply color conversion from RGB screen color to CMYK for printing, and it can even import color profiles, and it can display CMYK separations !

TIP: Adobe offers ZIP bundles of ICC color profiles for Mac and Windows, which include the popular USWebCoatedSwop.icc color profile.

OpenOffice Draw (Apache)

Over the years I have found Apache OpenOffice Draw to be an extremely handy drawing and illustration tool, and it is far more stable than it was in the past.

Given that it is free and that Adobe Illustrator is very expensive, this is a serious alternative for a wide range of drawing and illustration tasks. It is perfectly well suited to composition of business cards, brochures, posters etc. (as long as you are composing the more complex graphic elements in another tool). It can even handle SVG vector image import.

It is particularly good as far as text manipulation and precise page layout, layering, and element selection, positioning, and sizing are concerned, and in this respect it is much easier to use than say Pixelmator and The GIMP, which are really overgrown image manipulation tools, not drawing tools.

The Webel IT technology services business cards were composed using OpenOffice Draw:

GIMP graphics tool zone

The GNU Image Manipulation Program (GIMP) is a free, open source image and graphics editing tool with a flexible scripting facility. It is available for Linux, Mac and Windows.

The original 72dpi Webel animated GIF logo from about 2000 was developed
using some (rather keen) GIMP/Perl scripting. Watch out PIXAR !

The GIMP is not as full-featured as Adobe Photoshop, and not as easy to use as Pixelmator for Mac, and the interaction is not very Mac-like, but if you have patience you can certainly generate high quality graphics with it. It is definitely not very convenient as a drawing tool (it is not as handy as Adobe Illustrator or even OpenOffice Draw), it definitely belongs in the image manipulation family, although it has some handy pattern generators and some plugins that do help with drawing.

One of the best comparison articles I've found is GIMP vs Photoshop vs Pixelmator - Graphics Editor Shootout (Feb 2012) by Nathan Greenstein with some comparison tests and score tables.

I have found it over the years to be less than robust on Mac.

If you are going to install it on Mac, I highly recommend the extended bundles (offered per Mac OS X version) at GIMP on OS X rather than the stock GIMP builds at the main GIMP for Mac downloads area.

Pixelmator graphics tool zone

Pixelmator is a handy competitor to Adobe Photoshop. While not as full-featured, it is massively cheaper, and the feature gap is closing.

Certainly it seems to be more robust and easier to use in many respects than the free GIMP, although GIMP does seem to have more plugins and FX (like text-bevelling and bump-mapping).

One of the best comparison articles I've found is GIMP vs Photoshop vs Pixelmator - Graphics Editor Shootout (Feb 2012) by Nathan Greenstein with some comparison tests and score tables.

A printer-friendly 300dpi version of the Webel IT logo for business cards was composed in Pixelmator, with very slight bevel effect, and slightly more pastel coloured letters than in the original web version:

Mac OS X: audio engineering plugins

From Wikipedia: Virtual Studio Technology:

'Virtual Studio Technology (VST) is a software interface that integrates software audio synthesizer and effect plugins with audio editors and hard-disk recording systems. VST and similar technologies use digital signal processing to simulate traditional recording studio hardware in software. Thousands of plugins exist, both commercial and freeware, and a large number of audio applications support VST under license from its creator, Steinberg.'

'VST plugins generally run within a digital audio workstation (DAW), to provide additional functionality. Most VST plugins are either instruments (VSTi) or effects, although other categories exist—for example spectrum analyzers and various meters. VST plugins usually provide a custom graphical user interface that displays controls similar to physical switches and knobs on audio hardware. Some (often older) plugins rely on the host application for their user interface.

VST instruments include software simulation emulations of well-known hardware synthesizers and samplers. These typically emulate the look of the original equipment as well as its sonic characteristics. This lets musicians and recording engineers use virtual versions of devices that otherwise might be difficult and expensive to obtain.

VST instruments receive notes as digital information via MIDI, and output digital audio. Effect plugins receive digital audio and process it through to their outputs. (Some effect plugins also accept MIDI input—for example MIDI sync to modulate the effect in sync with the tempo). MIDI messages can control both instrument and effect plugin parameters. Most host applications can route the audio output from one VST to the audio input of another VST (chaining). For example, output of a VST synthesizer can be sent through a VST reverb effect.'

From Wikipedia: Audio Units:

'Audio Units (AU) are a system-level plug-in architecture provided by Core Audio in Mac OS X developed by Apple Computer. Audio Units are a set of application programming interface services provided by the operating system to generate, process, receive, or otherwise manipulate streams of audio in near-real-time with minimal latency. It may be thought of as Apple's architectural equivalent to another popular plug-in format, Steinberg's VST. Because of the many similarities between Audio Units and VST, several commercial and free wrapping technologies are available (e.g. Symbiosis and FXpansion VST-AU Adapter).'

'Mac OS X comes with Audio Units allowing one to timestretch an audio file, convert its sample rate and stream audio over a Local Area Network. It also comes with a collection of AU plug-ins such as EQ filters, dynamic processors, delay, reverb, and a Soundbank Synthesizer Instrument.

AU are used by Apple applications such as GarageBand, Soundtrack Pro, Logic Express, Logic Pro, Final Cut Pro, MainStage and most 3rd party audio software developed for Mac OS X such as Ardour, Ableton Live, REAPER and Digital Performer.'

From Quick Tip: How to Manage VST and AudioUnits Plugins in Mac OS X (2010):

'VST and AudioUnits (AU) are the two native plugin formats for Mac OS X. Although there are other DAW specific formats for plugins, VST and AudioUnits are more common and compatible across various DAWs like Cubase, Logic, etc. There is an abundance of VST and AU plugins for expanding your DAW and building your collection of effects. However, it can be difficult to know how to get those plugins running on your computer. Especially if they are free and do not come with installers or instructions. I’ll help you get those files in the right places and make them appear in your plugin stacks.'

'The plugin folder is nested in the Macintosh HD Library. There are usually a minimum of two Libraries on your Mac, one in Macintosh HD and another in your user account. You should only place the plugins in the Macintosh HD Library so that it can be accessed by all users on the computer. The usual location of the folder should be:

/Macintosh HD/Library/Audio/Plug-Ins/
$ ls -1 /Library/Audio/Plug-Ins/


How to Install VST Plugins

1. Unzip the downloaded file if it is an archive like .zip or .rar. You should only see a file with a .vst extension. This is the actual file required for the plugin.

2. Move the .vst file to the VST folder in your audio plugins folder.

3. If your DAW is running, close it and restart it. When your DAW starts up, it will rescan your plugins folder and detect your recently installed plugin.

How to Install AudioUnits Plugins

1. Unzip the downloaded file if it is an archive like .zip or .rar. You should only see a file with a .component extension. This is the actual file required for the plugin.

2. Move the .component file to the Components folder in your audio plugins folder.

3. If your DAW is running, close it and restart it. When your DAW starts up, it will rescan your plugins folder and detect your recently installed plugin.

Other Plugin Formats

You might come across another folder labelled VST3, this is for VST3 plugins which are not as common as of yet. They can be identified with the .vst3 file extension. MAS is used for MOTU Audio System. HAL is Hardware Abstraction Layer and you should not be needing to change anything there.'

Plugins for Audacity

Note: Audacity is a 32-bit application so won't see 64-bit versions of VST plug-ins, even on 64-bit operating systems.

From VST Plug-ins:

'In current Audacity (and legacy 1.3.8 and later), VST effects are displayed with full GUI interface (where provided by the plug-in), and without need of the VST Enabler. This has been made possible by use of an open source VST header.

When Audacity is first launched, an "Install VST Effects" dialogue will appear which lists VST plug-ins detected in the Plug-Ins folder inside the Audacity installation folder and in other system locations. Press OK to load the chosen plug-ins. Prior to Audacity 2.0.4, the scan happened automatically with no choice of which effects to load. Your VST effects will appear in the Effect menu, underneath the divider.

When you restart Audacity again it will reload the plug-ins it detected last session, as stored in the plugins.cfg file in the Audacity folder for application data. This avoids slowing down each Audacity launch by scanning for new plug-ins. So if you add more VST plug-ins later, you must go to the Effects tab of Audacity Preferences, check "Rescan VST effects next time Audacity is started ", then restart Audacity. If you subsequently remove any VST plug-ins, they will automatically be removed from the Effect menu after restart, without need for a rescan (as long as you are using 1.3.10 or later).'

From Audio Units:

'This page describes support for Audio Unit effect plug-ins in Audacity. Audio Units is a plug-in architecture developed by Apple and is only supported in Audacity 1.3.1 and later on Mac OS X.


Audio Unit support

Audio Unit (AU) support is available in Audacity 1.3.1 and later - Audacity scans for available AU plug-ins each time it launches. AU support is enabled by default, but it can be turned on or off by clicking Audacity > Preferences: Effects then under "Enable Effects", uncheck "Audio Unit". Restart Audacity for changes to take effect.

Audio Unit "MusicEffects" are supported in Audacity 1.3.14 and later. This class of Audio Unit supports audio input like pure "Effect" AU's but has the ability to use MIDI input to set effect parameters. Audacity doesn't yet accept MIDI input, so although MusicEffects should work fine as audio effects, parameters need to be set manually. Examples of MusicEffects are all those from DestroyFX, Ohm Force and SFXmachine, plus FXpansion Snippet, Tobybear MadShifta and u-he MFM2.

Like VST plug-ins in current Audacity, Audio Units display their full GUI interface by default, where one is provided. If interface difficulties arise, Audio Units can be limited to a tabular interface with sliders by unchecking the option "Display Audio Unit effects in graphical mode" at Audacity > Preferences: Effects. Once again, restart Audacity for changes to take effect.

.. You can find a useful list of third-party AU plug-ins (free and demo/paid-for) on Hitsquad.

To add new Audio Units (AU) plug-ins, place them in:




and restart Audacity. As always, ~ means your home directory. Audacity will not load Audio Unit plug-ins from the Audacity "Plug-ins" folder. '

From Nyquist Plugin-ins:

'Audacity supports Nyquist effects on all operating systems, and includes a number of Nyquist plug-ins. You can download additional Nyquist plug-ins, edit their behavior, or even write your own. Nyquist Plug-ins are merely plain text files which can be opened and studied using any simple text editor.


We host a large collection of Nyquist plug-ins for use in Audacity


On Windows and OS X, place new Nyquist plug-ins in the Plug-Ins folder inside your Audacity installation folder and restart Audacity. Your installation folder is usually under C:\Program Files on Windows computers, or under Mac Hard Disk > Applications on OS X.

On Linux, place new Nyquist plug-ins in one of the following locations:

- /usr/share/audacity/plug-ins if Audacity was installed from a repository package

- /usr/local/share/audacity/plug-ins if you compiled Audacity from source code

- ~/.audacity-files/plug-ins which is a per-user directory for which super-user privileges are not required (Note:

- The .audacity-files folder is not created during installation so must be created manually)

- in a Nyquist directory specified in the AUDACITY_PATH environment variable.

Restart Audacity then new plug-ins will be visible in either the Effect Menu, or sometimes in the Analyze or Generate menus. '

From Wikipedia: Nyquist (programming language):

'Nyquist is a programming language for sound synthesis and analysis based on the Lisp programming language. It is an extension of the XLISP dialect of Lisp.

With Nyquist, the programmer designs musical instruments by combining functions, and can call upon these instruments and generate a sound just by typing a simple expression. The programmer can combine simple expressions into complex ones to create a whole composition, and can also generate various other kinds of musical and non-musical sounds.

The Nyquist interpreter can read and write sound files, MIDI files, and Adagio text-based music score files. On many platforms, it can also produce direct audio output in real time.

The Nyquist programming language can also be used to write plug-in effects for the Audacity digital audio editor.

One notable difference between Nyquist and more traditional MUSIC-N languages is that Nyquist does not segregate synthesis functions (see unit generator) from "scoring" functions. For example Csound is actually two languages, one for creating "orchestras" the other for writing "scores". With Nyquist these two domains are combined.

Nyquist runs under Linux and other Unix environments, Mac OS, and Microsoft Windows.'

From Ladspa Plug-ins:

'LADSPA (Linux Audio Developers Simple Plugin API) is an audio plug-in standard originally developed on Linux, but which can be ported to Windows and Mac too. Audacity has built-in support for LADSPA plug-ins.'

Mac OS X: some audio engineering apps and tools

These are in addition to: FFmpeg: command line and GUI audio/video conversion tool: audio references

Most are known to run on Mac OS X Mountain Lion 10.8.5 as of 2013.


- FLAC tools: official command line tools for FLAC format.

- X Lossless Decoder (XLD): super little free GUI app for Mac OS X, can handle FLAC and ALAC and some other lossless formats, as well as converting from say FLAC to lossy formats like MP3 or AAC.

- Free Audio Converter (FREAC) GUI app: free audio converter and CD ripper. Features MP3, MP4/M4A, WMA, Ogg Vorbis, FLAC, AAC, and Bonk format support, integrates freedb/CDDB, CDText and ID3v2 tagging.

- Max: CD ripper and encoder that supports FLAC and some other formats.

Sound editors

- If I want to do anything exciting involving my own music I use the absolutely awesome Ableton Live for recording, editing, composition, and mastering. (BTW Ableton Live 9 supports multitrack recording up to 32-bit/192 kHz.) I can be engaged for professional audio services: visit Ableton Live (audio).

- Sometimes for post-processing or certain tasks I also use the audio editing in Final Cut, and I likewise offer professional media services for it: visit Final Cut video and audio editing and production.

But sometimes it is nice to be able to load a simpler audio editor for a quick fade-in/out or normalisation job, or just to make a quick recording.

- Audacity is a free, open-source, cross-platform audio editor for Mac, GNU/Linux Windows etc. It's not the world's best audio editor (especially not for MP3 or AAC because it imports, processes, then reexports with a small quality loss rather than say direct MP3 editing), but it has lots of FX and plugins and is sufficient for experiments, quick edits, and some post-processing, as well as wave analysis. As of Nov 2013 on OS X Mountain Lion 10.8.5, I find it far more stable than it used to be. Given that it's free, there's an awful lot that you can do with Audacity.

- To use Audacity with MP3 you will need to also install the LAME MP3 encoder, it's easy.

Internally Audacity works in uncompressed audio in 32-bit floating point by default, and offers up to 96kHz sample rate. You may simply import, edit, then export changes (losing edits), or save edited audio in its native AUP multi-file project folder format. In order to play the results in other programs, you must always export to another well-known format, and it supports nearly every format you will ever need.

- There is an unofficial Wave Stats plugin for Audacity that performs excellent wave analysis over regions of about 30s length, which is enough for you to explore the difference between dBFS RMS and max peaks.

To learn how to install Plugins for Audacity (and most other audio editors) on Mac OS X visit: Mac OS X: audio engineering plugins.

- From Rogue Amoeba for $32: Fission:

'Crop and trim audio, paste in or join files, or just rapidly split one long file into many. Fission is streamlined for fast editing. Plus, it works without the quality loss caused by other editors, so you can get perfect quality audio even when editing MP3 and AAC files. If you need to convert formats, Fission can do that too! You can rapidly export or batch convert files to the MP3, AAC, Apple Lossless, FLAC, AIFF, and WAV formats.'

I tried the free demo for file splitting on silences, not bad.

Here are some other editors I have not yet tried, but they might be worth a go:

- TwistedWave is available for Mac ($79.90), iPhone / iPad ($9.99) and online. TwistedWave for Mac is available as a fully functional 30 day demo. Can handle audio at a resolution up to 32-bit and 192 kHz sampling rate. Includes batch processing with silence detection for splitting long recordings into many files. Can perform pitch correction, pitch shift, and time stretch.

- NHC Software offer the Master's edition of WavePad for $59.95 (includes VST plugins and SFX library), however:

'A free version of WavePad audio editing software is available for non-commercial use only. The free version does not expire and includes most of the features of the normal version. If you are using it at home, you can download the free version here. You can always upgrade to the master's edition at a later time, which has additional effects and features for the serious sound engineer.'

Supports sample rates from 6 to 96kHz, stereo or mono, 8, 16, 24 or 32 bits.

There are dozens of other sound editors for Mac, but as far as I can tell, unless you are working on some real original music composition with something truly professional like Ableton, all you need is Audacity (free).


- Well obviously iTunes: plays most formats including WAV, AIFF, MP3 and AAC, likes compressed lossless ALAC, but does not play lossless compressed FLAC directly (yet). But that is not so bad because ...

- Fluke app: small OS X utility for listening to FLAC files within iTunes, without having to convert anything.

- QuickTime Player: although mainly known as a video player, is very useful for playing audio files (with a simple audio player GUI mode), and it also has a nice file info display with bit rates, sample rates etc. QuickTime is especially useful when you don't want to pollute your iTunes library with audio test files. Just right click and "open with .." then choose QuickTime Player instead of iTunes (or even set QuickTime Player as default for that audio file kind). However, as far as I can tell, QuickTime Player 10.2 still does not play FLAC.

- From Mac Software to play and convert FLAC:

The following software will play FLAC files without any requirement for modification - simply download, install and start using the current version.

- Cog:

- Play:

- VLC:

- Songbird:

- Bigasoft Audio Converter:

- From Audiofile Engineering for $US 19.99 Fidelia: Premium Music Player:

Fidelia is a high-definition audio player for sophisticated music lovers. With support for all contemporary audio file formats and an elegant interface that focuses exclusively on music, it gives users the power and the freedom to organize, customize and savor their digital music collection at the highest possible fidelity in any circumstance. If you've invested in premium audio hardware, you should have the best audio software.

Plays FLAC. Has adjustable real-time dithering.

- From Sbooth for $US 33 comes Decibel:

'Decibel is an audio player tailored to the particular needs of audiophiles. Decibel supports all popular lossless and lossy audio formats including FLAC, Ogg Vorbis, Musepack, WavPack, Monkey's Audio, Speex, True Audio, Apple Lossless, AAC, MP3, WAVE and AIFF. For lossless formats such as FLAC and WAVE, and for Ogg Vorbis and specially tagged MP3 files, Decibel supports gapless playback with seamless transitions between tracks. Decibel processes all audio using 64-bit floating-point precision, providing the highest possible playback quality for files sampled at all bit depths.'


Pro Level is a simple little $US 5 app with various VU-like digital monitors and some nice simple peak and clip hold settings, but you will need SoundFlower to shunt whatever stream your are targeting back through as an audio input source before it will see it (compare with Audio Hijack below, which you can also use to monitor system audio or any application's sound output directly).

Spectre: real-time Studio Multi-analyzer from Audiofile Engineering for $US99:

'Spectre is a multi-instrument real-time audio analyzer for Mac OS X. Designed in Cocoa from the ground up, Spectre proudly takes advantage of Quartz, OpenGL, CoreAudio, and other solid OS X interface features. Flexibility & Precision. Spectre focuses squarely on live audio analysis by offering 17 different multi-channel and multi-trace meters. Each meter can have any number of traces or indicators, and each trace can have it's own number of input channels, gain, mixing, filtering, ballistics and color (including transparency).'


- A likely "must have" for audio fun on a Mac is SoundFlower:

'Free Inter-application Audio Routing Utility for Mac OS X. Soundflower is a Mac OS X (10.2 and later) system extension that allows applications to pass audio to other applications. Soundflower is easy to use, it simply presents itself as an audio device, allowing any audio application to send and receive audio with no other support needed.

How To Use Soundflower

Soundflower presents itself as one of two audio devices (2ch / 16ch). The 2-channel device is sufficient for most situations. To send the output of one application to another, select Soundflower as the output device in the first application and Soundflower as the input device within the second application. If an application does not allow you to specify audio devices, you can make Soundflower the default input or output device inside the Sound panel in the System Preferences, or with the Audio MIDI Setup utility application. The 16-channel device is provided for more complex routing situations, and can be used with more than two applications simultaneously if the applications support audio routing to any channel, as Max/MSP does.'

But some of the functionality you might achieve with SoundFlower is more easily achieved out-of-the box with a good "hijacker".

Audio stream hijackers

"Exploring" and recording your (Mac) computer system's and applications' music sources (including online radio):

- Audio Hijack Pro (at around $US 32) is an absolutely super bit of software. You can record nearly any source (including any application) on your Mac, or full system audio. You can record Skype, Facetime, or anything you choose to "hijack", such as a particular web browser playing online radio. (Oops, I said it.) It has a very rich set of FX too, including tapping into all available VST and Apple FX, and you can customise nearly everything, including recording format, bit-rates, levels, schedule recordings, split recordings on-the-fly according to silence detection (with adjustable parameters). You can even use it to shunt audio around your system bus. Amazing !

- Also by Rogue Amoeba there is a new mini-version called Piezo, which unlike Audio Hijack Pro passes the restrictions to enter the Apple App Store. It enables you to record audio from any application, but you have restart the app every time after hijacking before a recording can start.

- To be fair I should also mention SoundTap (Mac and Win) from NHC Software, who also have a super kit of other audio apps. It is however not nearly as powerful as Audio Hijack Pro, but it's enough to tap a bit of your computer's sound quickly.

- And also Snowtape:

'Listen to internet radio. Record the music. Schedule radio shows. Edit songs and get album artwork. Export to iTunes.'

Hang on. Record the music ? From internet radio ? Ooh aah, that's naughty !

- And also Fstream for Mac. 'Listen to and record online radio easily'. Also available as an iPhone radio listening app.

No wonder so many online radio streams deliberately keep under 128kbps !

Some other audio apps and tools

- patch-based real-time audio and video synthesis. From the Max/MSP family. PureData is amazingly powerful and very clever. See also the Puredata synthesis zone for some examples. I am a huge fan of the PureData project; May Miller Puckette and the PureData/GEM community be blessed.

To see how I use PureData synthesise music and visuals from triaxial accelerometers to make real-time body music (gestural synthesis) see the Drancing project.

- MP3-Info is a very handy little app:

'MP3-Info is a clever companion that helps you organize your music collection. It is essentially a Plug-in for the Finder and iTunes. MP3-Info displays valuable information about audio files, such as their duration, the bitrate, important MP3-Tags, such as the artist, the title of the song, lyrics, cover art, and some more. That saves you a lot of time managing your song collection. It also shows these information for AAC files created by iTunes, and WAV, and AIFF files.'

- I haven't tried it yet, but AudioFinder sounds amazing. Can preview any audio file and give metadata and stats on any audio file direct in the Mac Finder.

- From TuneSweeper:

'Quickly find and remove all duplicates in your iTunes library. Remove missing iTunes files. Add additional music on your computer into iTunes.'

- FREE from AudioSlicer:

'AudioSlicer is a Cocoa GUI application for Mac OS X that finds all silences in an audio file and allows you to split it into several smaller audio files and to name/tag them properly. For now only MP3 is supported but other audio formats may be added in the future.'

Mac OS X: HOWTO adjust your system's sound quality, and record or find "high definition" audio sources

The following is at least applicable to Mac OS X Mountain Lion 10.8.5 in 2013.

Rule 1: there is no point having "high definition" audio resources if you don't
have your playback device set to handle high definition audio or if you don't
play them back on very good quality speakers or very good quality headphones.

What's the point of all those endless arguments about "best" and "minimal acceptable" sample rates and bit depths if your system is not even setup right to hear the difference ? It seems most Macs since many years before 2013 have supported at least 2-channel 24bit Integer sample bit depth at 96kHz sample rate, which is certainly higher definition than sleepy old stereo 16-bit 44.1kHz "CD quality", even if there is a lot of empirical evidence and psycho-acoustic science that proves you may never be able to notice the difference. So I give here some tips and links for checking "high definition" audio settings on your Mac.

But firstly, one of the curious things about the rough term "high definition audio" is that it is, well, not so well defined. You will often find it on the web loosely to mean 2-channel 24-bit 96kHz, while sometimes it is used for as low as 2-channel 24-bit 48kHz or even 44.1kHz, as long as 24bits per sample (per channel) are used. Some extreme audiophiles insist it is not true high definition unless it is at least 2-channel 24-bit 192kHz (and they probably also listen to ultrasonic "whistle music" at home together with their pet dogs and cats).

There is however a very clear definition of Intel High Definition Audio, from Wikipedia:

'Intel High Definition Audio (also called HD Audio or Azalia) refers to the specification released by Intel in 2004 for delivering high-definition audio ..

Hardware based on Intel HD Audio specifications is capable of delivering 192-kHz 32-bit quality for two channels, and 96-kHz 32-bit for up to eight channels. However, as of 2008, most audio hardware manufacturers do not implement the full high-end specification, especially 32-bit sampling resolution.

.. Mac OS X has full support with its AppleHDA driver. ..'

Well I'm not sure my beloved old MacBook Pro, Early 2008 17" can handle all that, but it certainly offers up to 2-channel 24-bit Integer 96kHz on built-in Microphone, built-in Input, and Built-in Output. You can check this for your system using a special application Utilities > Audio Midi Setup:

Whatever sample rate part of "Format" you choose there will also (on reload of the system info window) be reflected under About this Mac > System Report ...> Hardware > Audio > Devices, but it says nothing there about the sample bit depth:

Here you see I have now increased the sample rate for the output (only) to the maximum available for my system, 96kHz. The System Report is however strangely lacking in detail on the Intel High Definition Audio section (as is Apple's spec page for my Mac Book Pro 2008):

I have so far been able to find barely anything else concrete online about the audio hardware of any of my Mac machines, and in particular, I can't find anything about the credentials of the input ADCs - one good reason, especially when recording live music tracks, to instead use an external audio interface, say over USB or FireWire, with well known specifications instead, see discussion below.

High Definition Audio explained (2008) has a good discussion high definition audio as it is meant for Blu-Ray and HD DVD:

The Blu-Ray and HD DVD formats are capable of up to 48Mbps. Around 30Mbps of this transfer speed is reserved for video, leaving a sizeable chunk for (uncompressed) audio.

These audio streams can be sent to an AV receiver/amplifier as bitstream (encoded digital data) or PCM (essentially raw digital data.) Bitstreamed audio from a DVD, Blu-Ray or HD DVD disc needs to be decoded. This can sometimes be done by the player and output as PCM to the amplifier/receiver. More often than not though, the decoding is done by an AV receiver/processor. Regardless of which method you use, there is no difference in quality between PCM and lossless bitstreamed formats like Dolby True HD and DTS HD Master Audio. As a result, many Blu-Ray and HD DVD discs will offer both Dolby True HD/DTS HD Master Audio and (multi-channel) PCM soundtracks for the sheer convenience.

Along with the lossless Dolby True HD and DTS HD Master Audio formats, Blu-Ray and HD DVD offer Dolby Digital Plus and DTS HD High Resolution. While being a “lossy” format, these other two new standards offer benefits that Dolby Digital and DTS from DVD discs can’t such as higher sample rates.'

Of course HD DVD is now abandoned in favour of Blu-ray, for which Wikipedia explains (along with an excellent table of supported formats):

For audio, BD-ROM players are required to support Dolby Digital (AC-3), DTS, and linear PCM. Players may optionally support Dolby Digital Plus and DTS-HD High Resolution Audio as well as lossless formats Dolby TrueHD and DTS-HD Master Audio. BD-ROM titles must use one of the mandatory schemes for the primary soundtrack. A secondary audiotrack, if present, may use any of the mandatory or optional codecs.

Phew, well that made it easier. Basically, all Blu-ray players can at least support:

- Linear PCM (lossless): Max bitrate: 27.648 Mbit/s; Bits/sample: 16, 20, 24; Sample frequency: 48kHz, 96kHz, 192 kHz.

- Dolby Digital: Max bitrate 640 kbit/s; Bits/sample: 16, 24; Sample frequency: 48kHz; (Max 5.1 channels)

- DTS: Max bitrate: 1.524 Mbit/s; Bits/sample: 16, 20, 24; Sample frequency: 48kHz; (Max 5.1 channels)

In general, the higher sample frequencies are only available when no more than 6 channels are used.

So every BD-ROM player is required to at least support 24-bit at 192kHz (on 6 channels), which is higher than the 24-bit at 96kHz on 2 channels my old 2008 MacBook Pro can handle. It is however not as high as the 32-bit requirement for Intel High Definition Audio (see top).

Sourcing high definition audio (free)

Ok, now we know what high definition audio is (roughly) and how to ensure the Mac system audio is set for its audio highest definition capability, let's get some into our Mac. There are at least the following ways:

1. Record some yourself. Definitely the most fun and instructive, and the main subject for rest of this article. This way, as long as you do it right, because you get to explore the noise level, you know it is not only a high definition recording it is also a high definition source.

2. Get some free professional audio engineering test samples:

- Audio engineering test/sample file resources, and online generators and online audio tests

Especially useful are the generated ones, since you know there is not a lot of noise in them (or you know what kind of noise is in them).

Amongst the ones based on live recorded music, I found the free Steinway and Sons piano ones particularly interesting (as I am a piano player) and I confess I found it very hard in quality headphones to hear any difference between the CD quality 16-bit 44.1kHz WAV files and the 24-bit 96kHz WAV files.

3. Download free legal examples of high definition audio from all over the web (not necessarily professionally prepared audio test files, just any music). There is a good discussion of high definition audio resources here: How to find and play high-resolutions audio on the Mac, Jun 2011, by Kirk McElhearn. Includes also an excellent description of the inclusion of high resolution audio for sale on iTunes and other sites, and some of the lossless compression formats like FLAC (FLAC-HD) and ALAC often used to distribute them:

'Playing high-res files

Macs can natively support up to 24/96, played through iTunes or other software. However, without a couple settings tweaks, audio files with resolution higher than 16-bit/44.1kHz will automatically be downsampled to that resolution. So the first thing you need to do is set your sound output to 24/96. To do so, open Audio MIDI Setup, found in /Applications/Utilities. Select the desired output on the left, and then change the settings in the Format section on the right to 96000.0 kHz and 2ch-24bit.

Once you’ve made this change, you can play files at any resolution up to and including 24/96; lower-resolution files will actually be upsampled to 24/96 (which, unfortunately, won’t make them sound any better.)'

But you will never know for sure whether the actual music/sound source was in fact "high quality" and worth the high resolution audio treatment. The only way to be sure of that is to either record it yourself (very carefully), or get professional audio test files.

4. Steal illegal high resolution audio resources from all over the web (like high definition FLAC popular on torrent sites). I'll ignore this one. Besides, you can't be sure of what you get anyway !

5. Rip illegal high resolution audio resources from say Blu-rays at home. I'll ignore this one too. And you usually still can't be sure what the specs of the source behind the mastered audio put on the Blu-ray were, or the quality of the source, even if a "pro" did it for a major production house and a major entertainment distributor.

CAUTION: just because music samples are distributed using a high definition audio format does not mean the music is in fact "high definition audio". It is in some cases no more than a cynical marketing exercise. This is also sometimes true of so-called "high definition samples" offered to unsuspecting computer music composers who then work in a higher definition mode like 32-bit floating point (nevertheless of benefit from the point of view of internal processing, mixing, and FX application) and 96kHz sample rate in their DAW, but are in fact still working with music of no better than CD quality 16-bit 44.1kHz !
Professionally prepared audio test files are best for exploring high definition audio !

But it's fun trying to create your own, and one can learn a lot by doing it, so let's focus for the rest of this article on creating your own high resolution audio recordings.

Recording high(er) definition audio from live sources

Obviously, if the sound/music source is not clean with low ambient noise and your equipment is not clean with good specifications, there is little point. I found it however instructive to experiment with it even if there is some source or equipment noise clearly present to explore how the higher resolution recordings handle it.

Some remarks on audio interfaces

If you are recording it's no use having 24-bit depth at 96kHz on 2 channels if you can't get music into the machine at that quality. Many Mac model audio ports support both (through different connectors) analog via RCA and digital S/PDIF via TOSLINK with round mini-adapter (which is almost RCA shaped). From Wikipedia: TOSLINK:

'Also known generically as an "optical audio cable" or just "optical cable", its most common use is in consumer audio equipment (via a "digital optical" socket), where it carries a digital audio stream from components such as CD and DVD players, DAT recorders, computers, and modern video game consoles, to an AV receiver that can decode two channels of uncompressed lossless PCM audio or compressed 5.1/7.1 surround sound such as Dolby Digital Plus or DTS-HD High Resolution Audio. Unlike HDMI, TOSLINK does not have the capacity to carry the lossless versions of Dolby TrueHD and DTS-HD Master Audio.'

And from Wikipedia: S/PDIF:

'S/PDIF can carry two channels of uncompressed PCM audio or compressed 5.1/7.1 surround sound (such as DTS audio codec) with a maximum bandwidth of 3.072 Mbit/s per channel for a total of 6144 kbit/s; it cannot support uncompressed lossless formats (such as Dolby TrueHD and DTS-HD Master Audio) which require greater bandwidth like that available with HDMI or DisplayPort.'

But what's the use of Mac audio ports for recording from analog signals if Apple don't publish the A/D specs of specific machines ?

External audio interfaces

Another way of getting high quality audio sources for recording in - with known specs - is through an external audio interface to USB2/3, or FireWire400/800 (aka IEEE 1394). My MacBook Pro early 2008 has 3 USB2 ports and:

One FireWire 400 port at up to 400 Mbps

One FireWire 800 port at up to 800 Mbps

There have been lots of arguments online about whether PC cards are better than external interfaces, but with stable modern USB2/3 or FireWire and decent cables you will be fine (and for most Macs there is no choice, external it is). And there are lots of other nice things you can do with external audio interfaces. For example, some of them have very high quality mic preamps and stable phantom power. Some also have nice rerouting and inline FX capabilities.

I have an old M-Audio Firewire 410 (from about 2006), which also has excellent GUI software support, but the specs look a bit tired compared with modern interfaces:

M-Audio FireWire 410

• Dual low-noise mic/instrument preamps with gain controls, LED metering, phantom power and 66dB of available gain

• Two analog inputs (1/4" and XLR) and eight analog outputs on 1/4” TS jacks

• S/PDIF I/O on TOSLink optical or RCA coaxial connectors

• Supports sampling rates from 32KHz, up to 192KHz

• 2-in/8-out analog I/O at 24 bit resolution, up to 96KHz sampling rate

• 24-bit resolution playback at 192KHz sampling rate on outputs 1 and 2

• Two headphone outputs with assignable source and individual level controls

• Software-assigned rotary encoder for tactile control of monitor levels

• 1 x 1 MIDI I/O with hardware bypass switch for computer-independent operation

• Analog outputs support up to 7.1 surround using your audio software (your software must support surround outputs)

Frequency response 20-40kHz ± 1dB.

Signal to noise: –108dB

Dynamic range: 108dB (A-weighted)

• THD + N: 0.00281% @ 0dBFS

And from the box:

operating level: -10dB (unbalanced)

"Only" 96kHz sampling frequency is not bad (and matches the system audio capability of my old MacBook Pro), but many modern audio interfaces offer 192kHz. However at -108dB the noise level is a bit of a worry, although not the end of the earth, as explained in the summary of levels, loudness and noise at: A summary of a review of music levels for broadcasting, personal use, recording and mastering, including the new LOUDNESS measures.

BTW quoting SNR in negative (-)dB for such equipment is wrong according to this guide from RANE on audio specifications, and is often confused with EIN. Equivalent Input Noise or Input Referred Noise, which can be specified as, for example: EIN = -130 dBu, 22 kHz BW, max gain, Rs = 150 ohms.]

However compare 108dB SNR with some more modern audio interfaces/cards like the M-Audio Delta Audiophile 192 with an input SNR of 113dB, or the ASUS Xonar Essence STX with Input SNR of 118dB, and my old M-Audio Firewire 410 interface certainly seems well out of date. The modern cards and interfaces typically also have more generous frequency ranges from 10Hz to 90kHz (presumably for recording very big church organs and coyote howls).

Borrowing the excellent diagram from ZedBee's super article Digital recording rule of thumb, you can see that as long as you record (if using the 24-bit EBU Digital standard) with the RMS around -18dBFS, even 108dB SNR is not too bad (certainly good enough for even "high definition" home recording projects):

There is an excellent guide to Choosing A PC Audio Interface by Martin Walker from Sound on Sound mag from Nov 2004. The basic points are still relevant, with one of the major rules broken by my older M-Audio Firewire 410 (namely it uses only -10dBV consumer level voltage, not +4dBu pro level):

"Consumer & Professional

Many musicians are still confused about which interface input sensitivity and output level to use when faced with choices of [-]10dBV (consumer) or +4dBu (professional). It's easy to get bogged down in discussing millivolts and so on, but there are a few simple rules of thumb that should make everything easier to understand.
Always stick to the '+4' option if you can, since this generally results in lower noise levels. If you can't get high enough recording levels with '+4' input sensitivity on your interface, and there's no -10/+4 switch on the source gear, switch to '-10'. Similarly, stick with +4 output levels unless any connected gear can't cope with these higher levels, in which case revert to '-10'."

Aside: note carefully that these are in different dB scales, +4dBu and -10dBV (although product specs often state just +4dB or -10dB). From Understanding DB:

  • +4dBu equals 1.23 Volts RMS. Actually 1.2276 V
  • The reference level of -10dBV (0.316 V) is the equivalent to a level of -7.8dBu.
  • +4dBu and -10dBV systems have a level difference of 11.8 dB and not 14 dB. This is almost a voltage ratio of 4:1

Martin Walker seems to agree that one doesn't need the world's best signal-to-noise and dynamic range to make decent high quality recordings (although the needs of a true audio pro are more demanding):

'The most hotly quoted specification for any audio interface tends to be its dynamic range or signal/noise ratio. There's still a lot of confusion about these two terms, and this is hardly surprising considering each may be measured in a variety of ways. However, the way audio interface manufacturers measure them seems to be reasonably consistent, and using these particular methods the two figures also tend to be very similar with many products, which makes products that quote one or the other easier to compare.

In audio interface terms, Signal/Noise ratio compares the maximum signal level that you can send to the interface (ie. that which makes the input meters just register 0dB) with the background noise level when no signal is present. However, some crafty soundcard manufacturers realised early on that they could achieve amazingly good s/n figures by automatically muting the output in the absence of an input signal, so that its background noise level was significantly lower. The audio interface dynamic range measurement therefore measures the background noise level in the permanent presence of a low-level signal (generally a 1kHz sine wave at -60dBFS), which is subsequently notched out using a filter. Dynamic range is therefore a slightly more reliable real-world test. You may spot some cheap soundcards with significantly worse results for their dynamic range than for their Signal/Noise (S/N) ratio.


Both figures are generally measured via an 'A'-weighting network, which rolls off the noise either side of its 3kHz centre frequency, in line with the sensitivity of the human ear. In essence, a 'dBA' rating reflects more closely how annoying we will find the background noise, with low-level hums below 200Hz and whistles above 10kHz being less obvious than hiss between about 1kHz and 6kHz. A dBA rating is generally a few dBs better than a 'flat' measurement.

Despite the fact that most audio recordings still end up on a Red Book Audio CD at 16-bit/44.1kHz, most of us have abandoned 16-bit recording and playback in favour of the wider dynamic range possible with 24 bits. A typical soundcard will provide a maximum dynamic range of 96dBA at 16-bit, but well over 100dBA when using 24-bit, which allows us to worry less about taking our recordings to within a few dB of clipping, because the background noise levels are so much lower.

However, when comparing the dynamic ranges of different audio interfaces, don't lose sight of the signals you'll be recording. If, like me, you still record the outputs of various hardware synths, the chances are that they won't have a dynamic range of more than about 80dB. If you're capturing a live performance via a mic, the background noise level of that mic and its associated preamp may already be higher than that of the audio interface, especially since it's difficult to make recording areas really quiet without extensive soundproofing. After all, as Hugh Robjohns said in SOS September 2004: "In most public venues I find the ambient noise floor is typically about 50-55dB below the peak level of a modest orchestra, organ, or choral group".

So, while buying an interface with the lowest possible background noise is sensible, in the real world many musicians won't be able to hear any difference at normal listening levels between interfaces with a dynamic range of 110dBA and 120dBA. Moreover, I've recently spotted various musicians grumbling about the background noise levels of specific soundcards, when they were actually hearing digital nasties due to the the effects of a ground loop. As soon as they modified their wiring or introduced a DI box to deal with the problem, most were amazed at how quiet a background noise level of 100dBA was!'

Yep, that last one happened to me once, too. Buzz buzz, and it was it just a bad (dedicated) mic preamp with a ground loop. And Martin Walker takes away some more worry:

'It's also worth pointing out that switching to 32-bit recording and playback in your audio application won't result in an even larger dynamic range — the benefit of the 32-bit float format is massive internal headroom and no possibility of internal clipping when mixing together loads of tracks, but the interface will still have 24-bit converters on the input and output. Unless the world suddenly becomes a much quieter place, 24 bits will remain quite sufficient to digitise it.'

I found the following relevant, because I have an old 1993 Roland RD500 digital piano/organ with synth sounds (although I could not find out any specs regarding noise, or about how it produces its sounds or equivalent sample bit depths if it uses waveform reconstruction):

'Sample Rate Wars

While even budget audio interfaces are now beginning to feature 192kHz sample rates, there are still arguments raging on most audio forums about whether or not it's worth moving from a sample rate of 44.1kHz to 48, 88.2 or 96kHz. Many musicians stick to 24-bit/44.1kHz because they still create their music largely with hardware MIDI synths and soft samplers that themselves use 44.1kHz samples, so they see little point in moving higher, especially as they intend the final mix to end up on a 16-bit/44.1kHz audio CD. However, even those using electronic sources will probably find subsequent compression and peak limiting more accurate at higher sample rates, while EQ tends to sound far more analogue in nature and metering is more accurate. Those using soft synths that calculate or otherwise model their waveforms may also find they sound cleaner.

For live classical and other acoustic recordings I suspect most serious engineers now prefer 24-bit/96kHz, particularly if the final recordings are for DVD release at 48 or 96kHz ..

.. mainstream PC magazines may mark a particular review soundcard down if it doesn't offer a 192kHz sample rate, I personally consider this option a huge red herring in the case of most audio interfaces under £500. If you can hear the improvement, use 192kHz, but bear in mind that the rest of the signal chain needs to be of extremely high quality to really exhibit any benefit over 96kHz.

Remember, also, when choosing a sample rate for your projects, that at 192kHz every plug-in and soft synth you run will consume over four times as much CPU overhead, occupy more than four times the amount of hard disk space, and cut your potential simultaneous track count by more than a factor of four over 44.1kHz.'

Some more references

- Apple Tech Specs (with lookup against serial number).

- Apple: Mac Basics: Ports and connectors

- OS X Lion: Audio ports

- Audio Specifications: a super guide by RANE pro audio on interpreting the various audio specifications for devices, and how they should be measured and stated.

- Wikipedia: dog whistle

- 24/192 Music Downloads are Very Silly Indeed

- 24bit vs 16bit: the myth exploded

- The Emperor's New Sampling Rate, 2008 by Paul D. Lehrman.

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