WHAT IS APPLE DIGITAL MASTER?
The aim of this article is to better explain and understand what Apple Digital Masters for iTunes and Apple Music is. In this article, I'll explain how the conversion to lossy files affects the quality of a song and how Apple improve the preservation of audio quality.
First of all a quick answer
What is Apple Digital Master? Apple Digital Masters is a section of iTunes and Apple Music.
All the music in this section is badged and marketed as Apple Digital Masters. The final goal of this initiative is to give to artists, bands, producers and record labels the opportunity to deliver their music in the highest possible quality through iTunes and Apple Music. In order to do so, your master must be done by an approved Apple Digital Master Provider such as Master Your Track ...let's see more in-depth how it works from the technical point of view.
Apple Digital Master
For an average listener, the audio quality might not seem a big deal. However, for an artist, a producer, a recording, mixing, editing or mastering engineer having their work playback at the highest quality possible is essential as it is the result of their work and their fatigue. Not to mention that behind any single/EP/album there is also a quite consistent investment of money and energy.
The Apple Digital Master, in the past, called Mastered for iTunes, is an excellent opportunity for any artist to have their music digitally distributed at the highest possible resolution that Apple can offer. It also gives the opportunity of having the music badged and marketed as quality-checked music.
The only way to have your music accepted, badged and marketed as Apple Digital Masters is to have it mastered by and Apple Digital Masters Provider such as Master Your Track.
Digital Audio - Basic Understanding
PCM and AAC Encodings
PCM stands for Pulse Code Modulation, and it is a digital technology that allows storing audio files in a not compressed format. These formats take the extension of .WAV or AIF. Not being compressed these files are usually quite large.
With the development of digital players and the internet, streaming came up with the need for a lighter format.
MPEG (Moving Picture Expert Group) developed the MP3, a compressed file format which is much lighter than a .WAV of .AIFF. The downside of the MP3 is that it is compression affects the audio quality. (SEE THE ARTICLE: AUDIO FORMATS SHOUTOUT)
AAC stands for Advanced Audio Coding, and it has been developed by Apple to offer to its users the highest possible audio quality when using their services. This file format takes the extension of .M4A. If you have purchased music from iTunes, you're probably familiar with .M4A files.
Sample Rate and Bit Depth
With the diffusion of streaming platforms, the use of lossy files is needed due to their smaller and more manageable size. Lossy files like MP3, AAC, OGG compromise some information from the lossless PCM file in exchange for a smaller size. In other words, smaller files speed up the streaming process.
Lossy files are created starting from a PCM (.WAV or .AIFF files). In order to understand the conversion from a lossless to a lossy file, we need to have a basic understanding of how lossless files are created.
A PCM file (analogue to digital conversion) is created using sample rate and bit depth. The sample rate represents the number of samples taken per second. For instance, 44.1KHz means 44.100 samples in one second. The bit depth is the number of bits used by the computer to store and encode/decode a single sample. The computer sees these bits as a sequence of 0 and 1. (0011010101 - Have you watched Matrix? That is the code)
In the image below, we can see in red an analogue signal. In blue, are the digital samples. The number of blue dots expresses the number of samples taken or in other words, the sample rate. The position of the blue dots on the vertical axis expresses all the 16 combinations of 0 and 1 that we can have in a 4-bit file. (To know more about audio bit depth recommend the following article on Wikipedia)
So the sample rate is related to the frequency response, and bit depth is related to the dynamic or amplitude encoding, also known as quantisation (not to be confused with music notes quantisation).
Due to some limitations when converting a file from analogue to digital, we are going to have some quantisation distortion of the original signal.
The image below shows in green the original analogue signal. The black dots represent the signal quantised (sampled at a due sample rate and bit depth). In yellow is the rebuilt signal from the quantisation. In red, the difference between the original and the rebuilt signal introduced subtle distortion/noise.
Extra quantisation distortion is also introduced when decreasing the bit depth (for instance from 24 to 16 bits).
As you can imagine, the lower the bit depth, the higher the noise/distortion introduced. The conversion to a lossy file like an MP3 usually requires the reduction of the bit depth.
In order to mask and randomise these distortions at the mastering stage, some dithering is applied. Dithering is nothing more than a noise designed to cover up the distortion introduced by the reduction of bit depth, as you can imagine this results in an increased noise floor.
As previously said in order to reduce the file size and obtain a more manageable file for streaming the conversion to a lossy file deletes some information from the original and high-quality format.
Also, when creating a lossy file, and reducing the sample rate, some extra harmonics are generated. This problem is better known as aliasing, and the more you reduce the sample rate, the more this phenomenon becomes noticeable. (more about aliasing on Wikipedia)
From these points of view, AAC encoding has many benefits compared to MP3. The first one is that AAC is encoded using 24-bit quantisation. As a result, an AAC file has no added distortion/noise due to bit depth reduction, and it has a lower noise floor. As a consequence, no dither is needed. Sounds good, isn't it?
Another benefit of the AAC encoding lives in the process that Apple uses to convert the file from .WAV/.AIF to .M4A. The conversion happens it two steps: first, the. WAV/AIF is converted into a .CAF.
The .CAF is a 32-bit float file that results in a more accurate quantisation and consequently fewer inter-sample peaks. Moreover, during the AAC encoding, Apple uses an anti-aliasing filter to ensure that none of the harmonics generated during the downsampling is audible.
Another aspect to consider when mastering for Apple Music is the loudness. In fact, like many other streaming platforms, Apple Music normalises the audio in order to have a consistent volume through the songs.
Apple Music has created a unique software for loudness normalisation called Sound Check.
In other words, to make the listener experience as pleasant as possible and reproduce all the music at a similar volume, Apple normalises all the songs to a determined perceived loudness (-14 LUFS - Loudness Unit Full Scale - SEE THE ARTICLE: DB AND METERS...WHO ARE THOSE STRANGERS?).
What we need to consider as mastering engineers, mixers, and producers is that played at the same perceived volume more dynamic songs will sound much more impressive and pleasant than an over-compressed/limited one.
Apple is clear on what a mastering engineer must submit in order to conform to Apple Digital Masters standards. Apple not only gives the mastering engineers guidelines but also provides all the tools needed to verify the final files before the submission.
Such tools include:
Mastering Droplet: a standalone drag-and-drop tool that can be used to quickly and easily encode your masters using the Apple AAC encoder.
Afconvert: is a command-line utility that can be used to encode masters into AAC using the same commands Apple uses.
Afclip: is another command-line utility that can be used to check any audio file for clipping.
AURoundTrip: is an Audio Unit that can be used to compare the original source audio file to the encoded versions and can check for clipping, all in real time.
Audio to WAVE Droplet: automates the creation of audio files, in WAVE format, from any audio file natively supported on macOS.
You can find the tools at the following link: https://www.apple.com/itunes/mastered-for-itunes/
I hope this article has been helpful and if you have any questions, please don't hesitate to contact us.
To get your music on iTunes and Apple Music badged and marketed as Apple Digital aMaster, your mastering engineer must be on the Apple Digital Masters Provider List.
Master Your Track is on the
Apple Digital Masters Provider List.