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 harbour-lights.wav -vn -map_metadata 0 -ar 48000 -ac 2 -c:a libmp3lame -b:a 192k -f mp3 harbour-lights.mp3This takes the first 20 seconds of your first file (or the sample track), encodes it with MP3, AAC, Opus and Vorbis at five bitrates plus five lossless formats, then decodes every result and measures how much high-frequency detail survived. Nothing leaves your device.
Size and encode time are read from each output file; real bitrate is size divided by duration, which is why Opus and Vorbis land below the bitrate asked for: they treat it as a target. Detail kept is the energy above 16 kHz in the decoded result against the source, in dB. Lossy encoders drop the top of the spectrum first, so this shows how much detail each setting gave up. It is not a listening test, and quiet or band-limited sources will show smaller differences.
Each card sets the converter above to the format and quality that destination expects. Your files stay in the queue, so you can convert the same batch for two places in a row.
The last column fills in once the engine has loaded: it lists the encoders and containers the engine actually has, so the table shows what your browser can write right now. Click a row to pick that format. For how sample rate and bit depth work, see digital audio explained.