We begin with Pulse Width Modulation. In PCM, an analog audio signal is digitized by an ADC, which performs digitization by measuring its instantaneous voltage, and convert it to a n-bit integer that represents this voltage level, such as 16-bit, at a fixed sampling frequency, such as 44.1 kHz. This process is reserved to playback the audio. "To properly digitize and re-construct digital audio it's relatively important the sampling intervals be as accurate as possible between the recording process and when it's played back." If the timing is not exactly one sample per 1/44.1k second, the signal will be distorted, it's being frequency-modulated by the unstable timing and it produces additional unwanted distortion. Like any digital system, a clock signal produced by an electronic oscillator provides the timebase, also known as the reference clock. Usually, the ADC and DAC chips needs a clock frequency commonly used in digital circuits, from 5 MHz to 30 MHz, and the internal circuitry inside the chip use this frequency as the reference clock to derive all the timings needed by the chip, including the sampling or reconstruction of PCM. Same applies to all other digital encoding formats.

Ideally, the clock frequency should be as stable as possible. Unfortunately, all clock oscillators have inherent short-term instabilities, long-term instabilities, and non-zero temperature coefficient. Short-term instabilities, known as phase noise (in the frequency domain), or jitter (in the time domain) is a particular concern. As long as the audio system is properly optimized and characterized, jitter is not an issue. A good DAC will normally have a jitter around -90 dBc or lower, and it's probably negligible. NwAvGuy (an audio engineer known for his criticism of baseless audiophile practice), has a good explanation of clock jitter: https://nwavguy.blogspot.com/2011/02/jitter-does-it-matter.h...

But of course, some audiophiles want to power their Hi-Fi gears with the best oscillator available, so that it'll have the lowest jitter and the minimum absolute frequency error and temperature coefficient, even if the benefits is dubious to other people. And they realized that a rubidium frequency standard is the best clock oscillator they can find. First, it's possible to DIY. All you needed to do is understanding how the digital part of the Hi-Fi system works and find the crystal responsible for generating the system/ADC/DAC clock. You simply remove the crystal and inject an external clock signal from the rubidium standard to the chip. 10 MHz is a common output frequency from a standard oscillator, if the audio circuit also has a 10 MHz clock, the rubidium standard output can be used directly. If it's not, if the frequency radio of the two is a integer, it's possible to use a frequency multiplier or a divider to generate the needed clock frequency from the rubidium standard output. If it's not an integer, a PLL frequency synthesizer can be used.

Second, in an electronic lab, many gears have a "10 MHz Reference Input" port, their internal timings can be derived from the reference input, so that all signal generators, frequency counters, spectrum analyzer to a master clock oscillator (can be crystal oven, WWVB radio, GPS, or atomic) for consistency, so that all equipment's frequencies don't drift to different directions, all get a good clock with constant performance, and there won't be a disagreement between equipment. In some professional audio equipment for music production, it also allows an external reference to be used for the same purpose. One can simply plug the 10 MHz standard oscillator, including a rubidium standard, into the BNC connector of the audio gears.