![]() ![]() If your gain stage (pre-amps) are set properly everything should be brought up to 0dBu at the start of the channel strip. Therefor the + and - ranges tell you what is being done to your signal. Their 0 mark is defined by your input and nothing else.ĠdB on an audio control means the control is taking your input and passing it through with no gain or attenuation. ![]() Here is a cheat sheet I put together for myself on the topic.Īll of that being said, the dB markings on audio gear controls (not the meters) are arbitrary. As a reference a 16 bit recording has a noise floor at -98.09 dBFS. what changes is the number of steps BELOW that, you are just stretching a lower bit depth more to fill the full 1.23 Volt range of the +4 dBu output, and how high your noise floor will be on your final recording. 0 is set at the max since no matter what your bit depth, 0dBFS will be (should be) +4dBu on the output. This is used for digital audio and is most important at the analog to digital and digital to analog converter stages. dBSPL without a trailing letter implies no weighting has been applied.ĠdBFS = maximum dynamic range of the selected bit depth These are commonly noted simply as dBA, dBB, and dBC. It should be noted that there are three weightings (A, B and C weightings) for dBSPL that are used to compensate for the Fletcher-Munsen curve, or the variance in frequency response of human hearing at different levels. Used for sound pressure where 0dBSPL is the commonly recognized threshold of human hearing of 20 micropascals. This is used for consumer analog audio where consumer "line level" of -224 volts is represented as -10dBV "Line level" of 1.23 Volts RMS is represented as +4dBu. Here are the common ones you will see in audio nd their respective zero points. That tells you what units you are actually measuring, and what your x value is in the 20*log(x/y) calcualtion. The part that tells you your real unit is the letter or letters trailing the dB. Since the available "level range" is depending on the entire gear chain, a positive scale would give you different reference levels on different gear chains and in the digital realm it would mean your reference level would have a different value when you switch from 16 to 24 bit.ĭB in and of its self is not a unit but rather a for of notation, like the "10 to the power of" used in scientific notation. It tells you how far you are away from destroying the signal regardless of the actual voltage/power figures on input and output and lets you easily calculate gain/attenuation like "my signal peaks at -3dB, I still can make it 3dB louder if need be". In audio electronics, doing the same would have been quite impractical, there is no "absolutely quiet" in analog circuitry and you want to utilize the entire headroom of the signal chain and keep the signal as far above the noise as possible, so the reference level is the maximum level your device can digest without malforming it. Sound pressure level is measured in "positive dB" because the reference level 0dB is "absolutely quiet" for practical reasons (there is no "loudest noise ever" you could use as reference). I'm not really sure why the digital scale has to be negative, but that it how it is and there is no option to use another scale. So you want to have your analogue levels around this level while your recording so that you get the best sound out of your gear. This 0dbvu usually translates to around -18dbfs, but different converter designs can be calibrated to different levels. ![]() However unlike digital, it's alright if your signal goes over 0dbvu a bit because the distortion is usually gradual, and sometimes kind of nice sounding. At this point analoge gear is designed to have its sweetspot, where the signal is high above the noise of the circuit, but not so high that it starts to distort. This level is the zero of the analogue world, 0db vu. So you want to avoid hitting it so that your audio doesn't distort, and you need to keep at least 10db of a buffer away from it so that a sudden loud part doesn't ruin your recording.Īlso, though, the analog stages that your signal goes through, such as your preamps and the analog parts of your a/d converter are optimized to work at a certain level. This happens because digital audio describes the current position of a signal with numbers, and when you get up to 0 the numbers have run out, it's as high as you can go, so the same max number just keeps repeating and the peak of your waveform gets chopped clean off, which creates the sound of distortion. With digital audio, once a signal gets up to 0db it can't go past it, and the signal distorts. Fs means full scale (not sure why but there you go). ![]()
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