A dynamic range compressor reduces the difference between the loudest and quietest parts of audio. When the signal exceeds a threshold (in dB), the compressor reduces its gain by a ratio (e.g., 4:1 means 4 dB above threshold becomes 1 dB above threshold). The result: more consistent perceived loudness, with peaks tamed and quiet parts more audible.

Podcasts and interviews where guests speak at different volumes. Vocal recordings with inconsistent dynamics. Music meant for casual or noisy listening environments (cars, gym, headphones). Phone audio or video voice tracks. Anything that needs to sound consistent across the entire duration.

For podcasts: -20 dB threshold, 3:1 ratio. For vocals: -18 dB, 4:1. For aggressive loudness: -12 dB, 6:1. For mastering: -8 dB, 2:1. The defaults (-20 dB, 4:1) are a good starting point. Lower threshold = more compression. Higher ratio = harder compression.

Yes — over-compression sounds flat, lifeless, and "pumping" (you can hear the gain reduction breathing). For natural-sounding results, ratio under 4:1 and threshold not below -20 dB is safe. Always A/B compare against the original.

Normalizer (audio-normalizer tool) brings the entire file to a target loudness without changing the dynamics — quiet parts stay quiet, loud parts stay loud. Compressor reduces dynamic range — quiet parts get louder, loud parts get quieter. Often you compress first, then normalize.

Up to 200MB. Compression is a single-pass operation with low memory use, so most files process in a fraction of real-time.

Server-side tools use multi-threaded native FFmpeg running on dedicated CPUs with fast disks and parallel pipelines. Our engine is FFmpeg compiled to WebAssembly, which runs single-threaded inside your browser tab and has no access to native hardware acceleration. That makes browser-based jobs typically 3–8× slower than a server. The trade-off is total privacy: your audio file is never uploaded, never logged, and never stored — closing the tab erases everything from memory immediately. For most clips up to a few minutes the wait is small, and for sensitive recordings (voice memos, drafts, confidential meetings) the privacy gain is well worth it.

No. Everything runs entirely inside your browser tab using FFmpeg compiled to WebAssembly. The file is read into local memory only, processed in the same tab, and the result is offered as a direct download. Nothing is transmitted to any server, no account is required, no analytics are tied to your file, and closing the tab discards every byte from memory.

The file picker accepts audio inputs up to about 1 GB, which is well above what mainstream "free tier" online converters allow. The real ceiling is your device — everything runs inside your browser tab, which shares memory with the rest of the page. Most podcasts, songs, and voice memos sit comfortably under that limit even on a phone. If a very large lossless WAV or FLAC ever fails, trim it first or transcode to MP3 / Opus to bring the size down before re-running the tool.

MP3, WAV, OGG (Vorbis and Opus), FLAC, M4A (AAC), AAC, Opus, AIFF, and WMA all decode reliably via FFmpeg WASM. Output formats depend on the specific tool — most editing tools default to MP3 (universal) or WAV (lossless) but expose a format picker so you can pick the one that fits your downstream player or DAW.

Recent Chrome, Edge, Firefox, Safari, and other Chromium-based browsers all work. The tool relies on WebAssembly and SharedArrayBuffer, which require the page to be served over HTTPS with the right cross-origin headers — this site is configured correctly by default. On phones the same code runs but is slower than on a desktop because mobile CPUs are weaker.

No. The tool is completely free, requires no account, attaches no watermark, applies no usage caps, and shows no popup ads on your output. Because the work happens on your own device, there is no per-user quota for us to enforce — your hardware and browser memory are the only limits. The download is the file you would get from running FFmpeg locally, nothing more, nothing less.

For most music, 192 kbps MP3 is transparent — listeners cannot distinguish it from the lossless source. That typically gives a 6–10× reduction from CD-quality WAV. For speech-only content, 96 kbps Opus is transparent, giving a 12–15× reduction from WAV. Pushing further (below 96 for music, below 64 for speech) starts to introduce audible artefacts. The tool’s preview lets you A/B compare at different bitrates before committing.

Audio Converter focuses on changing format with quality preservation as the priority. Audio Compressor focuses on minimising file size, often by aggressively reducing bitrate, downsampling sample rate, or downmixing channels. Both tools use the same underlying encoders; they differ mainly in their default settings and in which controls they emphasise.

Yes, but with a caveat: re-encoding lossy audio adds another generation of compression loss on top of the original. A 320 kbps MP3 re-encoded to 128 kbps will be smaller, but it will sound noticeably worse than a 128 kbps MP3 made directly from the original lossless source. If you have access to the original, always re-compress from the lossless source rather than from a lossy intermediate.

Opus is a much newer codec (2012 vs MP3’s 1993) designed with the benefit of 20+ years of additional research. It uses smarter psychoacoustic modelling (predicting what humans cannot hear), better entropy coding, and adaptive switching between specialised modes for music and speech. The result is roughly 30–50% smaller files at the same perceived quality. The trade-off is compatibility — Opus is supported by all modern browsers and apps but not by some older devices.

For speech-only content (podcasts, audiobooks, voice memos), yes — downmixing to mono cuts the file size roughly in half with no useful loss. Most speech is recorded mono anyway and stored as stereo only because the recording chain happened to be stereo. For music, downmixing destroys the spatial information and is almost always the wrong move; reduce bitrate instead.

Transparent quality means a listener cannot reliably distinguish the compressed file from the lossless source in a blind A/B test. For most listeners on most equipment, 192 kbps MP3, 128 kbps AAC, or 96 kbps Opus crosses that threshold for music. For speech, the thresholds are roughly half those numbers. Audiophiles with high-end equipment may detect differences at higher bitrates; people listening on phones or laptops almost never can.

No — file-size compression (what this tool does) does not change loudness or dynamics. The audio comes out at the same volume as it went in. If you specifically want to change dynamics (compress loud peaks, raise quiet sections), that is a separate audio production task that requires a different tool, like a digital audio workstation.

Yes — drop a batch and the same compression settings apply to each file. Output files are named with the original base name plus a "-compressed" suffix so the originals are not overwritten. For mixed batches (some music, some speech), consider running them in two passes with format and bitrate tuned for each content type.