Audio Format Converter – MP3, M4A, Opus, FLAC and WAV Without Uploading
This audio format converter turns the audio files you already have into MP3, M4A (AAC), Opus, FLAC, WAV or an iPhone ringtone (M4R). Drop one song, or a whole album folder, choose a format, and download the result or a ZIP of every file. Decoding and encoding run inside your browser tab, so nothing is uploaded, and after the encoders load the first time the tool keeps working offline.
Containers versus codecs: why some conversions take a fraction of a second
A file name tells you the container; the sound inside is stored with a codec. An .aac file and an .m4a file can hold exactly the same AAC audio, and a WebM file from a browser recorder usually holds Opus. When the codec already matches the format you picked, at a bitrate no higher than the one you chose, the converter copies the audio packets into the new container instead of decoding and encoding them again. That keeps the quality identical and finishes almost instantly.
Lossy, lossless and generation loss
MP3, AAC and Opus are lossy: they throw away detail you are unlikely to hear to make files small. FLAC and WAV are lossless: they keep every sample. Converting one lossy format into another loses a little more each time, an effect called generation loss, so the tool starts the bitrate at roughly the quality of your source and marks it "≈ source quality". Picking a higher bitrate only makes a bigger file. Converting an MP3 to FLAC or WAV keeps it exactly as it sounds, but it can't bring back what the MP3 encoder removed.
How bitrate becomes file size
For lossy formats the size is simple arithmetic: bytes ≈ seconds × kbps × 125. A 4-minute song at 256 kbps is about 7.7 MB; at 128 kbps it is half that. WAV size is exact: seconds × sample rate × channels × bytes per sample, so CD-quality stereo (44.1 kHz, 16-bit) is about 10 MB a minute. FLAC depends on the music, so for a single file the converter compresses 30 seconds of it and scales the result, typically landing at 40–75% of the WAV size.
Sample rate, bit depth and why upsampling adds nothing
CD audio is 44.1 kHz and 16-bit; video and most phones use 48 kHz. Hi-res files go to 96 or 192 kHz and 24-bit. Raising the sample rate or bit depth of a file can't add detail that was never recorded, so the converter keeps the source's values by default. When you reduce a 24-bit file to 16-bit, samples are rounded without dither, which is inaudible for normal music but can matter for very quiet fades.
The frequency chart: spotting what an encoder removed and fake lossless files
Lossy encoders cut the treble above a fixed frequency, often around 16 kHz at 128 kbps and 19 kHz at 192 kbps, while natural recordings fade out gradually. The chart shows the average frequency spectrum of your file and, after converting, of the result, so you can see exactly what an encoder removed. The same "brick wall" in a FLAC or WAV is the classic sign of a fake lossless file made from an MP3 or AAC, and the tool warns you about it.
AIFF, WAV and tags
AIFF is Apple's twin of WAV: the same PCM audio with the bytes in the opposite order. The converter reads AIFF and AIFF-C of any length and keeps their tags. Tags and cover art carry across formats: ID3v2.4 in MP3, MP4 atoms in M4A, Vorbis comments in FLAC and Opus, and RIFF INFO in WAV. WAV's standard tags can't hold a cover, so an optional ID3 chunk is offered for that. For a single file you can edit the title, artist, album, year, track number and cover before converting.
Making an iPhone ringtone
An iPhone ringtone is AAC audio in an .m4r file, up to 40 seconds long. Pick a start time and length, add a short fade, preview the window and convert. Then drag the file onto the iPhone in Finder or the Apple Devices app.
Built with Mediabunny (MPL-2.0), the LAME MP3 encoder (LGPL), FFmpeg's AAC encoder (LGPL-2.1+) and libFLAC (BSD). For cutting audio use the Audio Trimmer, and to take the sound out of a video, the Video to MP3 Converter.