Lossy, lossless and uncompressed
Audio formats fall into three groups. Uncompressed formats such as WAV and AIFF store raw samples — large but immediately usable, which is why they remain standard in production. Lossless compression such as FLAC and ALAC reduces size by roughly 40 to 60 percent while reconstructing the original samples exactly. Lossy formats such as MP3, AAC and Opus discard information the ear is least likely to notice, achieving far smaller files.
Lossy encoders work through psychoacoustic modelling. A loud sound masks quieter sounds nearby in frequency and in time, so those masked components can be discarded without an audible difference. The encoder spends its bits where hearing is most sensitive, which is why a well-encoded 192 kbps file sounds far better than the number alone suggests.
For music, AAC and Opus both outperform MP3 at the same bitrate — Opus substantially so at low rates, which is why it dominates voice and streaming applications. MP3 persists on compatibility rather than merit, its patents having expired in 2017.
Transcoding compounds loss
Converting one lossy format to another decodes and re-encodes, and the second encoder treats the first one's artefacts as signal it must reproduce. Quality falls, and it falls further with each conversion. An MP3 converted to AAC and back is audibly worse than the original, even at a nominally higher bitrate.
Converting lossy to lossless does not restore anything. A 128 kbps MP3 turned into FLAC is a large file containing exactly the quality of a 128 kbps MP3 — sometimes described as an upscale, and it is purely wasted space. What discarded information was never stored cannot be recovered.
The rule is to keep an original in a lossless format and encode from it each time you need a different output. Where only a lossy source exists and conversion is unavoidable, use a generous bitrate to give the second encoder room, and accept that some degradation is unavoidable.
Choosing a bitrate and channel layout
For music at typical listening, 256 kbps AAC or 192 kbps MP3 is transparent for most listeners on most material, and going higher mainly consumes space. Variable bitrate encoding allocates bits according to complexity, so quiet passages use fewer and dense ones more — generally a better result than constant bitrate at the same average.
Speech has far narrower requirements. A podcast in mono at 64 to 96 kbps is entirely adequate, and Opus does well below that. Encoding a spoken-word recording in stereo at 320 kbps doubles the file for no benefit, since a single voice carries no stereo information.
Sample rate should generally be left alone. 44.1 kHz is the CD standard and 48 kHz is standard for video; both capture the full audible range, since the Nyquist limit puts the ceiling at half the sample rate. Converting between them requires resampling, which is a lossy arithmetic operation in itself — match your project's rate rather than converting repeatedly.