From studio WAV to a release-ready MP3.
A WAV straight out of a recorder or music app is raw and often too quiet or too loud for listeners. The file passes through three stages: set each one, and watch the waveform and size change.
Open songs, voice memos or old archives and save them as MP3, AAC, FLAC, Opus, WMA, AC3, AMR or any of 19 more. Pick a quality preset or an exact bitrate, convert the whole batch in one pass, then download each file or one ZIP.
ffmpeg -i input.wav -vn -map_metadata 0 -ar 48000 -ac 2 -c:a libmp3lame -b:a 192k -f mp3 output.mp3No compression at all: its size is pure multiplication. Samples per second × bits per sample × channels. Change any of the three and see how much space each one costs, and how a 256 kbps MP3 compares.
kbps = sample rate × bit depth × channels ÷ 1000
Loudness is measured in LUFS, an average of how loud something sounds to people, not its highest peak. Streaming services adjust every track to a target, so a master that's far louder only gets turned down, often losing punch. Match the target and it plays as you intended.
Constant bitrate (CBR) gives a quiet intro the same budget as a crashing chorus. Variable bitrate (VBR) saves on the simple parts and spends more on the busy ones: usually better quality at a smaller size. CBR still matters for a few old players and live streams.
The WAV is your master. Every other version should be made from it, never from an MP3.