WAV converter: what it does

This WAV converter takes an MP3, M4A, FLAC or OGG file and writes a standard 16-bit PCM WAV at the file's original sample rate and channel count, entirely in your browser. To convert a file to WAV, drop it, click Process and download; nothing is uploaded. You get the uncompressed format every DAW, sampler and mastering engineer can open, but the conversion cannot restore detail that a lossy encoder has already thrown away.

The status line shows the channel count, sample rate and length of what was loaded, then the size of the finished WAV in KB. The output keeps the original base file name with a .wav extension. For a different sample rate or bit depth, run the result through the sample rate converter or the WAV bit depth converter.

Convert MP3, M4A, FLAC or OGG to WAV: what changes

A WAV file is a container for raw PCM samples, so converting to WAV means decoding the source and writing the decoded samples to disk. You gain compatibility and a file that needs no decoder; you do not gain any audio that was not in the source. Lossy stays lossy.

Convert MP3 to WAV

MP3 removes audio the encoder judged inaudible, and the lower the bitrate the more it removes: most 128 kbps encodes are low-passed around 16 to 17 kHz, and a 320 kbps file keeps more top end but is still not the original. The WAV sounds exactly like the MP3 did on playback and is 4.4 times larger at 320 kbps or 11 times larger at 128 kbps (16-bit, 44.1 kHz, stereo). Convert when a sampler or an older application refuses MP3, not to improve the sound.

Convert M4A to WAV

M4A is a container, not a codec. Most M4A files hold AAC (iTunes purchases, iPhone voice memos), which is lossy like MP3, so AAC to WAV is a decode, not a restore. Some hold ALAC, Apple's lossless codec; those unpack to WAV with no loss, provided your browser can decode ALAC. Phone recordings are usually mono at 44.1 or 48 kHz; the readout tells you which.

Convert FLAC to WAV

FLAC is lossless, so FLAC to WAV gives back the exact samples that went in, as long as the bit depth matches. A 16-bit FLAC becomes an identical 16-bit WAV, roughly 1.5 to 2 times larger. A 24-bit FLAC is reduced to 16 bits by this converter, because it always writes 16-bit PCM; if you need to keep all 24 bits, use the WAV bit depth converter instead. Common reasons to convert: hardware samplers and some DAWs do not read FLAC, and a few plugins only browse WAV and AIFF.

Convert OGG to WAV

OGG files usually contain Vorbis or Opus, both lossy, and turn up in game audio and sample-site previews. Converting OGG to WAV writes the decoded audio as PCM, so it plays in players and samplers that have no OGG decoder. Quality is whatever the OGG had.

WAV formats: bit depth, sample rate and file size

WAV size is pure arithmetic: sample rate × bytes per sample × channels × seconds. The table gives the stereo size per minute for the formats you will actually meet. Mono is half. This converter writes the first column format (16-bit) at whatever sample rate the source had.

MB here means 1,000,000 bytes, the unit most operating systems show. For comparison, MP3 is 0.96 MB per minute at 128 kbps and 2.40 MB per minute at 320 kbps.

Worked example: a 3:30 track converted from MP3 to 16-bit, 44.1 kHz stereo WAV is 210 s × 44,100 × 2 bytes × 2 channels = 37.0 MB. The same length at 24-bit, 48 kHz would be 60.5 MB. To work out a track's length from bars and BPM before you estimate the file size, use the song length calculator.

Bit depth sets the noise floor, not the bandwidth: 16-bit gives about 96 dB of dynamic range, 24-bit about 144 dB, and 32-bit float adds headroom above 0 dBFS so an over-hot bounce does not clip inside the file. Sample rate sets the highest frequency the file can hold, at half the rate: 22.05 kHz for 44.1 kHz files, 24 kHz for 48 kHz files.

When you need WAV

DAW and sampler import

Modern DAWs decode MP3 and M4A on import by converting to PCM in the background, but many hardware samplers, drum machines and sampler plugins browse only WAV or AIFF. Converting once gives you a file that drags into any of them. If the sample has to sit in a project at a specific rate, run the WAV through the sample rate converter as well, so the DAW does not resample on the fly.

Mastering delivery

Mastering engineers want the mix as an uncompressed WAV or AIFF at the session's sample rate, usually 24-bit or 32-bit float, with no limiter on the master bus and a few dB of headroom. This converter writes 16-bit, so bounce that file from the DAW directly; where this page helps is turning a lossy reference the client sent into a WAV you can line up on the timeline. Our guide on how to master for streaming covers delivery specs and loudness targets.

Spotify, Apple Music and distributors

Distributors upload lossless files to the stores: WAV or FLAC, 16-bit or 24-bit, at 44.1 kHz or 48 kHz, and the store encodes its own streaming versions (Spotify streams Ogg Vorbis, Apple Music streams 256 kbps AAC). Apple's Digital Masters program asks for 24-bit files at the native session rate. An MP3 converted to WAV passes the file-type check but ships the MP3's artefacts to every listener, so convert for compatibility and keep the real master for release. Check the level before you deliver with the LUFS meter, and if a file just needs to hit a peak target without re-encoding, use the audio normalizer. Going the other way, for a small preview or demo, the audio to MP3 converter encodes a WAV to MP3 in the browser.

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