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A device The number of samples per unit of time is called an the <u>analog-to-digital convertersample rate</u> (ADC) turns analog electrical waves into digital signals, typically represented in kHz, which is the number of thousand samples per second. Conversely The sample rate determines the highest frequency that can be accurately represented, a which is called the <u>digital-to-analog converterNyquist Frequency</u> (DAC) turns digital audio into analog electrical signals, occurring at half the sample rate. With a 44.1 kHZ rate, the highest-representable frequency is 22.05 kHZ, which is roughly the maximum frequency that humans are capable of hearing. ADCTypical values are 16-bit/44.1 kHZ (used on typical consumer CD's ) and DAC's typically feature a low24-pass filter bit/96 kHZ (for commercial recording). Each frequency above the Nyquist frequency is digitized identically to remove high-frequencies that cannot be accurately represented by their maximum sample rateone below the Nyquist frequency. This Similarly, when the digital signal is at converted to analog, each frequency represented digitally will produce additional frequencies above the Nyquist frequency. Without filtering out such These "phantom" frequencies during the that are produced by analog-to-digital (A>D) and digital-to-analog conversion, certain frequencies may be incorrectly represented as lower frequencies, which are called <u>aliases</u> and their creation is called <u>aliasing</u>. Similarly, the digitalA>D and D>A converters use low-pass filters to-analog conversion will produce remove frequencies above the Nyquist frequency which are aliases of lower frequencies. They should be filtered out for accuracy (however, it shouldn't matter if the Nyquist frequency exceeds the range of human hearing)to prevent aliasing.
→Basic Principles
The most basic way to think of digital audio is like animation or video. By playing a number of individual "frames" quickly enough, it gives the appearance of motion. In the case of audio, the frames are called <u>samples</u>. Unlike video, however, digital audio is not an illusion - the digital representation of the sound is capable of storing all the information contained in and can be converted back into identical sound waves, at least theoretically.
The amount of sound information a digital audio signal can represent is determined by the bit depth and the sample rate. <u>Bit depth</u> is the number of bits per sample. Each sample represents the amplitude of the sound wave at that point in time. The higher the bit-depth, the more distinct values are possible for each sample, allowing more and more accuracy in recording. The number of samples per unit of time If the bit depth is called significantly low, the <u>values for each sample rate</u>, typically represented in kHzwill be quite "off" from the analog wave. When such digital audio is converted back to analog, which is the number of thousand samples per second. The sample rate determines resulting waveform will have a significant difference from the highest frequency that can be accurately representedoriginal. The This is called <u>Nyquist Frequencyquantization distortion</u> is 1/2 the sample rate , and is the highest-pitched frequency that it can be accurately representedresult in increased noise and added frequencies in addition to simply loss of fidelity. A sample rate of 44.1 kHZ means the highest-representable frequency Sometimes this effect is 22.05 kHZ, which is roughly the maximum frequency that humans are capable of hearing. Typical values are 16-bit/44.1 kHZ (used on typical consumer CD's) and 24-bit/96 kHZ purposefully (for commercial recordingusually in electronic music), referred to as bitcrushing.
While digital audio is theoretically capable of perfectly representing analog sound, in practice, devices can inaccurately measure amplitude or mis-time a sample. The quality of such devices is paramount to accurate recording and reproduction of sound. Commercial audio production uses bit depths and sample rates higher than the final mastered product to ensure any error is minimized. While there is no need for frequencies between 22.05 kHZ and 48 kHZ (as no human can hear them), the higher sample rate prevents signal degradation as digital audio streams are manipulated and mixed together.
What has driven digital audio to completely replace analog formats in both the production and consumer side of the music industry? The best answer is its ability to be copied infinitely without any loss of quality. Recorded takes can be copied, sliced up, rearranged, manipulated, mixed-down, etc. without worry of any loss of quality. Similarly, multiple tracks can be mixed down with no quality loss. Record companies can distribute the master copy of an album to consumers without any degradation as it is transferred to its a physical medium. It has also reduced costs. The price of transferring a compressed audio file from a server to a home computer is miniscule. Transferring while transferring data to and from magnetic tape reels, or creating plates to press vinyl records is not. Similarly, rather than having to record each take in a recording studio on expensive high-quality magnetic tape that would degrade in quality with erase and rewrite operations, recording audio digitally onto a hard drive is virtually cost-less. There's also no time costs in terms of rewinding, erasing, etc. Computers - computers manipulate digital data almost instantaneously.
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