Audio Compressor

MP3, WAV, OGG, M4A, AAC, FLAC, and any format supported by the browser Web Audio API.

WAV (PCM). For smaller compressed formats like MP3, use FFmpeg after downloading.

Peak normalization scales the audio so the loudest peak reaches 0 dBFS without clipping, maximizing perceived loudness.

Yes, the Audio Compressor is 100% free with no registration, no hidden fees, and no usage limits. All processing happens locally in your browser, ensuring complete privacy.

Absolutely. The Audio Compressor processes everything client-side in your browser. No data is uploaded to or stored on any server. Your content remains private on your device at all times.

Yes, the Audio Compressor is fully responsive and works on smartphones and tablets. You can use it on any device with a modern web browser -- no app download required.

No account or registration is needed. Simply open the Audio Compressor in your browser and start using it immediately. There are no sign-up walls or usage restrictions.

Simply enter your input in the provided field, adjust any settings to your preference, and the tool will process it instantly. You can then copy the result to your clipboard or download it.

The Audio Compressor works in all modern browsers including Chrome, Firefox, Safari, Edge, and Opera. For the best experience, use the latest version of your preferred browser.

Because the output is uncompressed WAV (PCM), the file size follows a simple formula: sample rate x channels x (bit depth / 8) x duration. That means cutting the sample rate cuts the size proportionally. Dropping from 44100 Hz (CD quality) to 22050 Hz roughly halves the file; going to 16000 Hz is about a third of the original, and 8000 Hz (telephone quality) is under a fifth. The trade-off is high-frequency detail: each halving discards the top octave of sound, which speech barely uses but music depends on. Combine a lower sample rate with mono and 8-bit depth and the savings multiply together. This tool shows the estimated output size and percentage saved before you commit, so pick a sample rate and watch the badge update to hit your target.

For spoken word, 16000 Hz mono is the sweet spot. Almost all the energy in human speech sits below 8 kHz, and a 16000 Hz sample rate captures frequencies up to 8 kHz cleanly, so dialogue stays crisp and intelligible while the file shrinks dramatically. Voice is also rarely stereo in any meaningful way, so converting to mono halves the size again with no real loss. If you want a touch more warmth for narration or an interview, 22050 Hz mono still sounds natural and stays small. Reserve 44100 Hz for music or anything with cymbals, strings, and air you want to preserve. Turn on Normalize first if your recording came in quiet, then set 16000 Hz, enable Convert to Mono, and download a lean WAV ready for upload.

This happens because the tool outputs uncompressed WAV, while your source was probably already a compressed format like MP3, M4A, or AAC. Those codecs squeeze audio into a fraction of its raw size, so a 3 MB MP3 expands to tens of megabytes of raw PCM when decoded. If you then pick a sample rate higher than the original, or keep full 44100 Hz stereo 16-bit, the WAV the tool writes is simply bigger than the small compressed file you started with. Choosing a sample rate above the source never adds real quality either, since the missing detail cannot be recreated. The estimate badge warns you with a Larger than input label when this occurs. To genuinely go smaller than an MP3, lower the settings here, then re-encode the downloaded WAV with FFmpeg using the command shown in the tool.

They control two separate dimensions of the sound. Sample rate is how many times per second the audio is measured, which sets the highest frequency that can be reproduced; halving it from 44100 to 22050 Hz removes the top end and roughly halves the size. Bit depth is how precisely each of those measurements is stored. Staying at 16-bit keeps a wide, clean dynamic range, while dropping to 8-bit stores each sample in half the data, making the file about 50% smaller but adding audible hiss in quiet passages because there are fewer steps to describe soft sounds. They stack, so 8-bit at a low sample rate compounds both the savings and the quality loss. For most clips, lower the sample rate first and keep 16-bit; reach for 8-bit only when size truly outweighs fidelity.

Converting to mono merges the left and right channels into one, which immediately cuts a two-channel file's size by about 50% because you store half as many samples. Whether you lose anything audible depends entirely on the content. Plain narration, voice memos, and most podcasts are effectively mono already, so collapsing them is free size with no perceptible change. Music with deliberate panning, stereo effects, or a wide spatial mix is different: mono flattens that width and can dull the sense of space, so keep two channels there. The conversion itself is clean, not a quality hack, it simply removes the separation between channels. If you are unsure, listen to whether anything moves between your speakers. For spoken content, switch on Convert to Mono here to shave the size with no real trade-off.