Consider the signal chain of a live audio system, from a singer to a PA.
The singer produces sound pressure waves into a mic. These pressure waves are measured in dB SPL, and the microphone's specifications will tell you its sensitivity, maximum sound pressure level, frequency response, on- and off-axis response, and so on. An engineer can look at all these specs to choose a mic based on the environment, how much background noise there is and where it's coming from, the quality and character of the singer's voice, etc.
The microphone will produce a voltage, which is measured in dBV or dBu. This voltage travels to a preamplifier, which boosts the signal to a nominal level by applying a gain (in dB, with no units since it's just a dimensionless ratio between input level and output level) and converts it to a digital signal -- measured in dBFS ("decibels full-scale", signal level relative to the maximum level that can fit in the bit-width of the digital signal).
Note that all the different signals going into the preamps have wildly different voltage levels -- a kick drum produces much more sound energy than a singer. And the ratio between SPL and voltage is not fixed; different mics have different sensitivities, and an active guitar amp will generate a signal many orders of magnitude stronger than a passive microphone. For some instruments like synthesizers, "SPL" isn't even a concept that makes sense because the sound is produced entirely electronically, rather than by capturing a mechanical wave. So, the preamps are configured to normalize all the incoming signals to one nominal level for processing.
After the preamp, the signal goes through a digital signal processing chain. Most of these processing steps will affect the level of the signal, and the amount is measured in dB (without units, since this is a dimensionless ratio between input level and output level. Remember, we're dealing with a fully digital waveform, so there is not even a physical measure of "loudness" or "signal strength" that can apply to the signal at this point.)
The signals from all the different sound sources are mixed together, and the mixer's per-channel volume faders are marked in decibels -- usually from +10 to -infinity, with 0 corresponding to "unity gain" (which means the output signal should have the same intensity as the input signal). Again, no units because we aren't measuring a physical quantity, just how much the mixer should change the intensity of the signal.
Finally, the signal is converted back to an analog voltage (measured in dBu) and sent to a power amplifier. The power amplifier applies an adjustable gain, and outputs a signal measured in watts (dBm) for sending to the PA system. You might then walk around the room with an SPL meter to ensure the sound is at a safe level; the value you measure will depend on the frequency and directional response of the speakers, as well as how far away from the speaker you're measuring.
Throughout this process we had signals in the form of sound pressure, voltage, digital samples, and wattage, at power levels ranging from microwatts to kilowatts. Even a simple physical quantity like sound pressure for a single signal is not straightforward -- how far are you measuring it from, and are you talking about SPL before the microphone or after the speaker?
The fact that a single unit is used for all of these different purposes is a feature, not a bug. If my preamp is close to clipping and I turn it down by 3 dB, how much do I need to turn up the gain at my compressor to compensate? Easy -- 3 dB. If we used more "appropriate" physical units at each step of the signal chain, it would be impossible to determine this number quickly or in your head; and would involve taking into account even more factors I haven't mentioned (like input impedances). The fact that decibels are scaled differently depending on whether you're working with voltage or power means that 1 dB corresponds to the same change in loudness everywhere without having to remember whether it's 1 dB of voltage or power.