I've been puzzled by something for a while now. There are questions at the end, but the preamble is my attempt to think my way through some of the background info. There may be plenty of mistakes along the way, so feel free to point them out. That said, here we go: My understanding is that dBFS refers to the measurement of the levels of a digital signal, such that 0 dBFS represents the highest possible level that a signal can be before you experience digital clipping, i.e. where you run out of bits to store the information about the waveform and end up just storing anything over 0 dBFS as if it were the same as 0 dBFS. That might be a weird way of putting it. For example, in a 16 bit recording every sample is a 16 bit value where each of the 16 bits can be 0 or 1, and so 1111 1111 1111 1111 is the highest value you can capture (which would be 2^16 or 65536, right? Although I’m not sure what units to attach to that number). If if the voltage level of the incoming analog signal is higher than that, the A/D converter is only going to be able to send the data as 1111 1111 1111 1111 because there are no more bits available to capture the additional information. This "clips" the signal digitally. And it sounds like crap. Once inside my DAW (Cubase), the signal is processed at 32 bit floating point, so there's all kinds of headroom available, 16 extra bits in this case, to store additional data and do processing through software EQs and reverbs and whathaveyous. On playback or when mixing down to a stereo wav at 16/44, either my DAW or my interface or both are going to turn that 32 bit data back into 16 bit data. Although I suppose I could record at 16 bit, process in Cubase at 32bit FP and then mix down to a stereo 24 bit file, although I have no idea why I would. But it's possible. Anyway, I've read somewhere that in 16 bit recording, 0 dBFS corresponds to 96dB. And in 24 bit, it's 144dB, so that's the amount of dynamic range (?) available at each bit depth. And I think I understand that while 0dBFS is a level you don't want to exceed unless you're trying to make something sound clipped, we use RMS values to measure loudness since RMS levels give a closer approximation of how the human ear actually hears sounds or loudness in general over time, not just split second levels or peaks that may come close to or exceed 0dBFS. So far so good? I usually record at 24 bit using a Mackie Onyx 1640 interface. I have two questions: (1) if 24 bits allows 144dB of dynamic range, what is that 144dB? I mean: what kind of dB’s are these? Not dBFS. dBu? dBm? According to Wikipedia, there are about 25 different kinds of dB. So what does that 144dB actually mean? And (2) My Onyx manual says the following things: a. Dynamic Range: >110 dB (mic in to main out) b. Input gain control range: 0dB to +60dB <- the mic generates a dB level itself prior to hitting this preamp, I assume; especially if I used an outboard preamp that created even more signal on the way into the board, right? c. Maximum voltage gain: mic in to main output: 80dB So question 2 is what are those dB's referring to? dBu?