Frequently asked questions
What does Mastered for iTunes mean?
Mastered for iTunes is a mastering procedure to ensure that there are no clipped samples in the encoded audio files. Clipping can occur for two different reasons:
1) Intersample overloads. A sharp peak in the audio signal can have its True Peak between two samples at maximum amplitude. As the signal is converted back to the analog domain, the analog signal would have a peak that is higher than 0 dBFS and is therefore clipped.
2) In an encoded signal using perceptive coding such as mp3, aac or ogg, signal components that we supposedly do not hear are removed – not encoded – in order to save bits. As a result, the time signature is altered and may result in overloads although the original audio in PCM format is not overloaded.
When mastering for iTunes, the audio files are encoded using the AAC codec in the OS X operating system and are then checked to make sure that no samples are clipped and that there are no intersample overloads.
What is MP3 and AAC?
MP3 and AAC are two encoding schemes for audio based on the properties of human perception. One such property is that high levels of low frequency sound makes higher frequency sounds with lower levels inaudible. This is one example of a masking effect and as a result of that, fewer bits can be used to represent the audio signal.
When a PCM signal is converted to MP3 or AAC, supposedly inaudible parts in the original signal are removed and sometimes we refer to this as destructive or lossy encoding. An audio signal encoded as MP3 or AAC cannot be restored to the original PCM signal.
What is PCM?
When an analog audio signal is converted to a digital signal, a sample is taken from the analog signal 44100 times per second for CD resolution and 96000 times or more per second for High Definition (HD) recordings. For each sample taken, the signal voltage is measured and converted to an integer number. This number – the sample value – represents the voltage at one time instance with some precision, referred to as the resolution and depends on the number of bits in the integer word.
The CD resolution is 16 bits according to the Red Book standard and because of this, the allowable input voltage range for the AD-converter (typically ±12.3 V for a professional AD-converter) is divided into 65 536 (2^16) voltages that can be represented by the integer sample values. The discrepancy between the true analog voltage and the voltage represented by the sample is an error, which is sometimes referred to as noise. Under some assumptions, a maximum signal-to-noise ratio (SNR) can be calculated and is typically given roughly as 6 dB per bit, resulting in 96 dB for a Red Book CD.
Nowadays, most AD-conversions are made with 24-bit converters with 16 777 216 discrete voltages and a maximum signal-to-noise ratio of 144 dB. This is actually a much larger signal-to-noise ratio that what we can achieve with analog circuits, so in effect the (SNR) is determined by the analog circuits (typically microphones and microphone amplifiers) preceding the AD-converter.
A PCM-encoded signal is thus a stream of digital numbers represented as 16, 24 or 32 bit integers or 32-bit floating point numbers. The floating point numbers typically have a 24-bit mantissa (determining the SNR) and an 8-bit exponent (determining the dynamic range). This is the format used in e.g. WAV files. PCM stands for Pulse Code Modulation.