Thank you SteveG. I mostly use Adobe for ID/AI/PSD, so I'm afraid I might not follow some of your audio terminologies. If you don't mind me asking, can you tell me step-by-step on how to achieve what Bob and you are talking about? (thank you ryclark as well) Am I correct to assume I'm already working on 32-bit floating point (see attached image)? The project I'm working on is to mix 3 hours of music for a coffeeshop. Each individual track is of different style, and comes at unmatched volume. My goal is to find a solution to match the volumes of all tracks evenly, so when it's played at the coffee shop, each track to each track the volume's difference is not jarring. Your advice to either use Normalize or Dynamic Processing – that's for each individual music clip correct? I wouldn't do me any good to mix the multitrack session down then apply normalize/d.process afterward, correct? Please tell me if I'm wrong: from what I've read online, I should use normalize on tracks where in the waveform's peak level is well below 0db (i.e. overall gain): However, the online resource said: Peak normalization to 0 dBFS is a very bad idea for any parts to be used in a multi-track recording. It may not clip by itself, but as soon as you add any extra processing or play tracks simultaneously your DAW or Plug-ins may overload. This subject comes under ‘gain staging’, a big subject to cover in the future. So what is the best practice in my case? Or does the 32 bit floating point basically bypass this issue altogether? On Dynamic Processing: I should use Dynamic Processing on clips that I want to lessen the dynamic range (meaning the loudest part will get less loud, while softer part gets louder, correct?) Is this then recommended to use on music genres with wide sonic range such as classical music, where some parts will be very quiet, thus get drowned out by ambience noise of the coffee shop? To quote Bob: First, if you're working in 32 bit Floating Point, you can lower the levels later (using things like Dynamics Processing and/or Normalise) and you won't get any clipping. "Later" here means when? After mix down? If that is so, it's ok for the multitrack to clip then adjust later? Please pardon my lack of audio knowledge! Btw, I am currently manually adjusting the levels like this:
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