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Drop audio files to change bitrate
or click to browse — MP3, WAV, AAC, M4A, FLAC, OGG, OPUS, WMA, AIFF, AMR, CAF
Select audio filesChange the bitrate of any audio file — right in your browser. This free Audio Bitrate Changer lets you increase, reduce, optimize and standardize bitrate across MP3, AAC, M4A, FLAC, OGG, OPUS and WAV, with CBR/VBR/ABR control, smart recommendations, quality analysis and batch processing. Nothing is uploaded and there’s no watermark.
Increase, reduce or standardize bitrate with one-click presets or a custom value.
Choose constant, variable or average bitrate encoding for the best size-to-quality balance.
Analyzes each file and suggests the ideal bitrate for voice, podcasts or music.
See loudness (LUFS), true peak, dynamic range and clipping before and after.
Convert dozens of files at once with a managed queue and one-click ZIP download.
MP3, AAC, M4A, FLAC, OGG, OPUS, WAV and more — change format and bitrate together.
Live estimates of file-size difference, compression ratio and bandwidth saved.
No upload, no account, no watermark. Everything runs in your browser, forever free.
Drag and drop your files, or click to browse. Add one or many — batch processing is built in.
The tool reads each file’s format, current bitrate, sample rate, channels and loudness automatically.
Choose a goal (reduce, match, increase or optimize), a preset, or set a custom bitrate and CBR/VBR/ABR mode.
Change the output format, sample rate, channels, normalize loudness or adjust volume.
Click Convert. Compare original vs result, then download each file or all of them as a ZIP — no watermark.
Audio bitrate is the amount of data used to store one second of sound, measured in kilobits per second (kbps). A 320 kbps MP3 uses far more data per second than a 64 kbps file — and that extra data preserves more detail. Bitrate is the single biggest lever on the trade-off between audio quality and file size.
An audio bitrate changer re-encodes your file at a new bitrate so you can shrink a track for sharing, standardize a podcast feed, or re-export at a higher ceiling. This tool does it for free, entirely in your browser, with full control over format, encoding mode, sample rate and channels.
For lossy formats like MP3, AAC and Opus, a higher bitrate generally means better quality and a larger file. Below about 128 kbps you start to hear artifacts in music; 192–320 kbps is transparent to most listeners. Speech and podcasts stay clear at much lower bitrates — 64–96 kbps mono is plenty.
One honest caveat: increasing the bitrate of an already-compressed file cannot recover detail that was discarded when it was first encoded. Going from a 128 kbps MP3 to 320 kbps makes a bigger file, not a better-sounding one. The real wins come from reducing bitrate to save space, or re-encoding a lossless source (WAV/FLAC) to a high-quality lossy file.
Changing bitrate with this tool takes seconds. Add your file, choose a target, and download — the conversion runs locally so nothing is uploaded.
CBR (Constant Bitrate) uses the same bitrate for every second of audio. It’s the most compatible choice and makes file sizes predictable — ideal for streaming, broadcast and older players.
VBR (Variable Bitrate) spends more bits on complex passages and fewer on quiet ones, giving the best quality for a given file size. ABR (Average Bitrate) targets an average while varying moment to moment — a middle ground. For most music, VBR at V2 (~190 kbps) is the sweet spot; for podcasts and maximum compatibility, CBR is the safe pick.
The right bitrate depends on the content and where it will be played. Use lower bitrates for speech and sharing, higher for music and archiving. Match the sample rate to the source (44.1 kHz for music, 48 kHz for video/studio) and keep stereo for music, mono for voice.
Podcasts and audiobooks are spoken-word, so they compress beautifully: 64–96 kbps mono keeps voices clear while keeping episodes small enough to stream and download quickly. Add loudness normalization so every episode plays at a consistent level.
Music needs more headroom: 192 kbps is a great default, 256–320 kbps for premium quality, and lossless FLAC for archiving or mastering. Streaming platforms typically deliver 128–320 kbps AAC, so exporting at 256 kbps AAC/M4A matches what listeners hear.
Inputs: MP3, WAV, AAC, M4A, FLAC, OGG, OPUS, WMA, AIFF, AMR, CAF and more — plus video files, whose audio track is extracted. Outputs: MP3, AAC, M4A, OGG, OPUS, WAV, FLAC, AIFF, ALAC and AC-3.
You can change the bitrate within the same format (e.g. a 320 kbps MP3 down to 128 kbps) or convert format and bitrate at once (e.g. FLAC to a 256 kbps M4A). Lossless outputs (WAV/FLAC/AIFF/ALAC) use bit depth and compression instead of a bitrate.
Your audio never leaves your computer. All decoding, analysis and re-encoding happens inside your browser tab with WebAssembly FFmpeg, so there is no upload, no cloud copy and no third party handling your files.
Only your chosen settings can be saved locally for convenience. The audio itself is held in memory and released as soon as the conversion completes.
| CBR | VBR | |
|---|---|---|
| File size | Predictable & fixed | Varies with content |
| Quality per MB | Good | Best |
| Compatibility | Highest (all players) | Very high (modern players) |
| Best for | Streaming · broadcast · voice | Music · archiving |
| Seeking accuracy | Exact | Slightly less precise |
| Use case | Recommended | Why |
|---|---|---|
| Voice / audiobook | 64 kbps mono | Clear speech, tiny files |
| Podcast | 96–128 kbps | Standard spoken-word quality |
| Standard music | 192 kbps | Great everyday quality |
| High-quality music | 256 kbps | Premium streaming quality |
| Maximum MP3 | 320 kbps | Best lossy quality |
| Archiving / mastering | FLAC (lossless) | No quality loss at all |
Standardize every episode to 96 kbps mono, normalized, for fast, consistent downloads.
Re-export masters to 320 kbps MP3 or 256 kbps AAC for distribution and demos.
Compress long recordings to compact 64 kbps mono files that stream smoothly.
Shrink a music library to fit more tracks on your device without re-ripping.
Drop a song to 96–128 kbps so it sends instantly on WhatsApp, Telegram or email.
Conform audio to 48 kHz and a target bitrate to match your video project.
Your audio is decoded and re-encoded locally with WebAssembly FFmpeg — it never touches a server.
Nothing is saved in the cloud. Only lightweight settings stay in your browser.
In-memory buffers are released the moment a conversion finishes or you close the tab.
Served over HTTPS with strict, cross-origin-isolated headers for fast, safe processing.
Kilobits of data per second of audio. Higher = more detail and bigger files.
Constant Bitrate — the same bitrate throughout. Predictable size, most compatible.
Variable Bitrate — bits allocated by complexity. Best quality per megabyte.
Average Bitrate — targets an average while varying moment to moment.
Samples captured per second (44.1 kHz = CD, 48 kHz = studio/video).
Loudness Units Full Scale — a perceptual loudness measure used for normalization.
Lossy (MP3/AAC) discards data to shrink; lossless (FLAC/WAV) keeps it all.
Re-encoding audio from one format/bitrate to another.
Convert audio between WAV, MP3 and more.
Shrink audio file size with full controls.
Merge multiple audio files into one.
Trim and cut a single audio file.
Convert any audio or video to MP3.
Reduce video file size online.
Reduce the size of any file fast.