Your files stay on your Mac.
Convertessa hands WAV files to FFmpeg's LAME encoder and writes MP3s on your Mac at the quality you set. Sample rates MP3 cannot express are resampled first, tags with an ID3 equivalent come across, and mono stays mono rather than being doubled into stereo. A folder of session bounces, interview takes or CD rips converts as one queue instead of one file at a time.
WAV is a container with almost nothing in it: a RIFF header, then raw PCM samples exactly as the recorder or the DAW produced them. That is why one stereo minute at 44.1 kHz and 16-bit costs about 10 MB, and why a field recorder can fill a card in an afternoon. MP3, encoded here by FFmpeg driving LAME, turns that minute into roughly a tenth of the size by discarding what a psychoacoustic model predicts you will not hear. Unlike the file size, that decision cannot be undone later.
Three properties of the source decide whether the encode is uneventful or surprising:
Metadata is the quieter loss. WAV keeps notes in RIFF LIST and INFO chunks, and broadcast WAVs add a bext chunk holding the originator and a sample-accurate timestamp used to sync sound to camera. MP3 carries ID3 tags, which have artist, title and album but nowhere to put a timecode. If you are converting production sound, the MP3 is a listening copy — the WAV stays the master.
Last, gapless. Every MP3 opens with a short run of encoder-added silence, which players only skip if they read the LAME header. Rip a continuous live set track by track and you may hear a tick between songs on anything that ignores it.
Drag in a folder of .wav bounces, rips or takes. Files queue individually, so a 96 kHz interview and a 44.1 kHz music bed sitting side by side each convert on their own terms.
Pick MP3 and set quality. What suffers first at low settings is predictable: cymbals, applause, room reverb and sibilance, the dense high-frequency material the psychoacoustic model has the hardest time approximating.
Convert, then archive the WAVs somewhere safe. Decoding an MP3 back to WAV restores the file size but none of the discarded audio, so the original recording remains your only master copy.
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It is resampled to a rate MP3 can express, normally 48 kHz. MP3 as a format only defines 32, 44.1 and 48 kHz for full-rate streams, so there is no way to keep 96 kHz in the container. In practice this is not the loss people fear — the material above 24 kHz was inaudible anyway. The audible change comes from the lossy encoding, not the rate conversion.
Two reasons, both structural. The encoder inserts a short priming run of silence at the start so its filter bank has something to chew on, and the audio is packed into fixed-size frames, so the end is padded out to fill the final frame. The LAME header records both so a good player can trim them. Total added time is a few tens of milliseconds — invisible for a podcast, audible as a tick between gapless album tracks.
No. WAV stores markers in a cue chunk, often with labels in an adjacent list, and MP3 has no equivalent — ID3 was designed for library metadata, not editorial positions. Chapter-style markers are lost, and so is anything a DAW wrote into a custom chunk. Export a marker list from your editor before converting if you need those positions later.
Classic WAV cannot exceed 4 GB, because RIFF stores chunk sizes in 32 bits — roughly six hours of 44.1 kHz stereo, or far less multitrack. Long-form recorders sidestep this by writing RF64 or Wave64 files, often still named .wav. FFmpeg reads those, so a long lecture or concert recording converts normally. A truly malformed oversized WAV, written by a recorder that crashed mid-take, is the case that fails.
Point the CLI at the folder and give it a destination:
convertessa ./takes/*.wav --to mp3 --quality high -o ~/Music/Takes
Named presets map to numbers, so high is the same as --quality 90 and balanced is 80. Filenames are kept, extensions swapped, and nothing is written over your originals.
Yes, sensibly so. A mono WAV encodes as a mono MP3 rather than being duplicated into two identical channels, which keeps the file roughly half the size for the same fidelity. Speech is the ideal case for MP3 in general: a narrow spectrum, no dense cymbal wash, and a psychoacoustic model that has an easy job. Interview archives shrink dramatically with no audible difference.
You can produce a WAV from an MP3, but it is not a recovery. Decoding restores the file size and the sample rate; the frequency content the encoder threw away does not return, and the decoded file inherits every artefact plus the encoder's padding. Treat MP3 as a distribution format in one direction only, and keep the WAV if the recording is irreplaceable.