Bitrate and sample rate
Sample rate is how many snapshots per second of the sound; bitrate is how much data per second the exported file spends.
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What it is
Sample rate is how many measurements per second are taken of the sound wave (44.1kHz means 44,100). Bit depth is how precise each measurement is (16-bit, 24-bit). Bitrate is a different animal: how much data per second the file spends. Every audio file has one, including PCM and lossless formats (48kHz, 24-bit, stereo WAV works out to 2,304 kbps, no choice involved). Lossy codecs like MP3 and AAC are just where the number gets pushed in your face as a quality dial, because there you are picking how much to throw away.
Practical defaults: record and edit at 48kHz / 24-bit if the audio meets video, 44.1kHz for music-only delivery. Deliver lossless masters as 16-bit or 24-bit WAV. For distribution, 128kbps AAC is fine for spoken word, 192kbps to 256kbps for music, and a mono podcast can live happily around 96kbps. Higher sample rates buy you headroom for pitch shifting and heavy processing, not audible detail: 48kHz already captures everything up to 24kHz, above human hearing.
Two real traps. First, mismatched sample rates: dropping a 44.1kHz file into a 48kHz timeline without conversion plays it at the wrong speed and pitch, or drifts against picture. Second, generation loss: re-encoding an MP3 into another MP3 stacks artifacts every time, so always keep a WAV master and export lossy copies from it rather than from a previous export.
Ask AI for it
Export this finished mix in three delivery formats from the same 48kHz 24-bit WAV master: a 48kHz 24-bit WAV archive master, a 48kHz 16-bit WAV for video editorial, and a 192kbps VBR AAC for web streaming. Never re-encode from a lossy file, resample with a high-quality converter, and report the source sample rate, bit depth, and channel count of every input file before converting.