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CALCPARK

Audio Converter

Convert audio between MP3, AAC (M4A), OGG Vorbis, Opus, WAV and FLAC in your browser, several files at once, with a bitrate choice for lossy formats. Nothing is uploaded.

Files stay on your device
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How to use it

  1. Drop one or more files (or a video whose audio you want).
  2. Choose the target format. For sharing pick MP3; for Apple devices AAC; for the smallest speech files Opus; for archiving FLAC.
  3. Set the quality: High (MP3 256 kbps, AAC 192 kbps) is transparent for most music; Low is enough for voice.
  4. Press Convert. Download each file, or all of them as one ZIP.

Example

Three WAV recordings from a field recorder (44.1 kHz stereo, 10 MB per minute, 4 minutes each) converted to MP3 at High: each becomes about 7.5 MB, the batch takes 10–15 s, and “Download all” gives one 23 MB ZIP. The same recordings as Opus at Standard (96 kbps) are 2.9 MB each and still sound clean on headphones; as FLAC they shrink to roughly 55% of the WAV size with no loss at all.

What each conversion runs

Every file is one FFmpeg command inside a Web Worker; nothing is sent anywhere:

ffmpeg -i input.ext -vn -map 0:a:0 <encoder options> output.ext
  • MP3: -c:a libmp3lame -b:a 128k | 192k | 256k | 320k, ID3v2.3 tags.
  • AAC (M4A): FFmpeg’s native aac encoder, -b:a 96k | 128k | 192k | 256k, +faststart so the file streams. (The browser build has no libfdk-aac.)
  • OGG Vorbis: -c:a libvorbis -q:a 3 | 5 | 7 | 9– VBR quality levels that average roughly 112, 160, 224 and 320 kbps.
  • Opus: -c:a libopus -b:a 64k | 96k | 128k | 160k; the encoder resamples to 48 kHz as the format requires.
  • WAV: -c:a pcm_s16le – 16-bit PCM at the source sample rate. FLAC: -c:a flac -compression_level 5.

-vn drops cover art and video; -map 0:a:0 takes the first audio stream. Title, artist and album tags are carried over where the target container supports them. Channel count and sample rate are preserved except for Opus (48 kHz). Files larger than a few hundred MB can exceed the 2 GB WebAssembly memory limit.

Sources: FFmpeg – audio encoders (libmp3lame, aac, libvorbis, libopus, flac), Xiph.org – Vorbis quality levels and bitrates, ffmpeg.wasm documentation.

Which format should I choose?

MP3 when the file must play anywhere, including old car stereos. AAC/M4Afor iPhones, iTunes and video soundtracks – slightly better than MP3 at the same bitrate. Opus for voice memos, podcasts and messaging: 64 kbps Opus sounds like 128 kbps MP3, but some older players cannot open it. OGG Vorbis for games and open-source software. FLAC to archive CDs or masters without loss at about half the WAV size; WAV only when an editor or sampler insists on uncompressed PCM.

Questions

Does converting MP3 to WAV or FLAC improve the sound?
No. The detail removed by the MP3 encoder is gone; a lossless copy only takes more space. Convert lossy to lossless only when a program requires that container.
Can I convert M4A or WMA to MP3 here?
Yes. FFmpeg decodes AAC, ALAC, WMA, AIFF, AMR, AC-3, Opus, Vorbis and more; choose MP3 as the target. Protected files (DRM) from stores cannot be decoded.
What happens to the cover art and tags?
Text tags (title, artist, album) are copied where the target format supports them; embedded cover images are dropped because the picture stream is excluded with -vn.
Are my files uploaded?
No. Conversion happens in your browser with FFmpeg compiled to WebAssembly. The only download is the 32 MB engine from the unpkg CDN on first use; afterwards the browser cache serves it.