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The simplest way to change the duration or pitch of an audio recording is to change the playback speed. For a digital audio recording, this can be accomplished through sample rate conversion. When using this method, the frequencies in the recording are always scaled at the same ratio as the speed, transposing its perceived pitch up or down in ...
Mastering, a form of audio post production, is the process of preparing and transferring recorded audio from a source containing the final mix to a data storage device (the master), the source from which all copies will be produced (via methods such as pressing, duplication or replication).
For example, Compact Disc Digital Audio and Digital Audio Tape systems use different sampling rates, and American television, European television, and movies all use different frame rates. Sample-rate conversion prevents changes in speed and pitch that would otherwise occur when transferring recorded material between such systems.
FLAC (/ f l æ k /; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation.
Using this technique, the audio data is stored as a sequence of fixed amplitude (i.e. 1-bit) values at a sample rate of 2.884 MHz, which is 64 times the 44.1 kHz sample rate used by CD. At any point in time, the amplitude of the original analog signal is represented by the density of 1's or 0's in the data stream.
High-resolution audio (high-definition audio or HD audio) is a term for audio files with greater than 44.1 kHz sample rate or higher than 16-bit audio bit depth. It commonly refers to 96 or 192 kHz sample rates. However, 44.1 kHz/24-bit, 48 kHz/24-bit and 88.2 kHz/24-bit recordings also exist that are labeled HD audio.
ALAC supports up to 8 channels of audio at 16, 20, 24 and 32 bit depth with a maximum sample rate of 384 kHz. ALAC data is frequently stored within an MP4 container with the filename extension.m4a. This extension is also used by Apple for lossy AAC audio data in an MP4 container (same container, different audio encoding).
Another type of normalization is based on a measure of loudness, wherein the gain is changed to bring the average loudness to a target level. This average may be approximate, such as a simple measurement of average power (e.g. RMS), or more accurate, such as a measure that addresses human perception e.g. that defined by EBU R128 and offered by ReplayGain, Sound Check and GoldWave.