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Various halvings and doublings of 44.1 kHz are used – the lower rates 11.025 kHz and 22.05 kHz are found in WAV files, and are suitable for low-bandwidth applications, while the higher rates of 88.2 kHz and 176.4 kHz are used in mastering and in DVD-Audio – the higher rates are useful both for the usual reason of providing additional ...
Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let () be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring the value of the continuous function every seconds, which is called the sampling interval or sampling period.
If the ratio of the two sample rates is (or can be approximated by) [A] [4] a fixed rational number L/M: generate an intermediate signal by inserting L − 1 zeros between each of the original samples. Low-pass filter this signal at half of the lower of the two rates. Select every M-th sample from the filtered output, to obtain the result. [5]
The DVD format uses the 48 kHz sampling rate, and its doublings. In digital audio, 48,000 Hz (also represented as 48 kHz or DVD Quality) is a common sampling rate. It has become the standard for professional audio and video. 48 kHz is evenly divisible by 24, a common frame rate for media, such as film, unlike 44.1 kHz. [i]
Digital Audio Tape (DAT or R-DAT) is a signal recording and playback medium developed by Sony and introduced in 1987. [1] In appearance it is similar to a Compact Cassette, using 3.81 mm / 0.15" (commonly referred to as 4 mm) magnetic tape enclosed in a protective shell, but is roughly half the size at 73 mm × 54 mm × 10.5 mm.
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.
The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample rate must be at least twice the bandwidth of the signal to avoid aliasing.
For example, uncompressed songs in CD format have a data rate of 16 bits/channel x 2 channels x 44.1 kHz ≅ 1.4 Mbit/s, whereas AAC files on an iPod are typically compressed to 128 kbit/s, yielding a compression ratio of 10.9, for a data-rate saving of 0.91, or 91%.