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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 ...
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 selection of the sample rate was based primarily on the need to reproduce the audible frequency range of 20–20,000 Hz (20 kHz). The Nyquist–Shannon sampling theorem states that a sampling rate of more than twice the maximum frequency of the signal to be recorded is needed, resulting in a required rate of greater than 40 kHz.
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.
Sample-rate conversion including upsampling and downsampling may be used to change signals that have been encoded with a different sampling rate to a common sampling rate prior to processing. Audio data compression techniques, such as MP3 , Advanced Audio Coding (AAC), Opus , Ogg Vorbis , or FLAC , are commonly employed to reduce the file size.
[a] But in signal processing, decimation by a factor of 10 actually means keeping only every tenth sample. This factor multiplies the sampling interval or, equivalently, divides the sampling rate. For example, if compact disc audio at 44,100 samples/second is decimated by a factor of 5/4, the
The higher sample rates impose less restrictions on anti-aliasing filter implementation which can result in both lower complexity and less signal distortion. Work done in 1981 by Muraoka et al. [23] showed that music signals with frequency components above 20 kHz were only distinguished from those without by a few of the 176 test subjects. [24]
For instance, to implement a 24-bit converter, it is sufficient to use a 20-bit converter that can run at 256 times the target sampling rate. Combining 256 consecutive 20-bit samples can increase the SNR by a factor of 16, effectively adding 4 bits to the resolution and producing a single sample with 24-bit resolution. [3] [a]