Audio-Kompressor

No. The compression runs in a version of FFmpeg compiled to WebAssembly inside this page, so the file is processed on your own machine. That matters for voice notes, interviews, and recordings of meetings, which is most of what people compress.

For music, 192 kbps is high quality and 128 kbps is good enough for most listening. For speech — voice notes, interviews, podcasts — 64 kbps is clear and 32 kbps is still intelligible while being tiny. Going above 320 kbps in a lossy format gains nothing worth the size.

Yes, but only downwards. Re-encoding at a lower bitrate makes the file smaller and slightly worse, because lossy compression discards detail every time. Re-encoding at a higher bitrate than the original just wastes space — the detail is already gone, and the tool tells you when the result is no smaller.

Opus, by a clear margin at low bitrates, and it is excellent for speech. AAC (M4A) beats MP3 below about 128 kbps. MP3 is the least efficient of the three but plays on absolutely everything, which is why it is the default.

It merges the two channels into one, which roughly halves the data needed at a given quality. For a voice recording that is usually invisible; for music it flattens the stereo image, so keep stereo there.

Yes — enter a target size in MB and the tool works out the bitrate needed to land there, instead of you guessing a bitrate and checking afterward. It is most useful when there is a hard limit to hit, like an email attachment or an upload cap.

It reduces how many times per second the audio is sampled, which shrinks the file further on top of the bitrate. 22.05 kHz is a safe cut for speech and barely noticeable; 16 kHz saves the most but starts to sound thin, so keep it for voice notes rather than music.

About 300MB. The whole file is held in the tab while it is processed, so beyond that a browser is likely to run out of memory. Longer recordings are better split first, or handled by a desktop tool.