By Tim King · Updated
Audio to WAV Converter
Convert MP3, M4A, FLAC or OGG to 16- or 24-Bit WAV, Resampled or Mono
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How the Audio To WAV works
Convert audio to WAV without installing anything. Drop in an MP3, M4A, AAC, FLAC, OGG or any other file your browser can decode and export it as a plain PCM WAV at 16 or 24 bits. Keep the source sample rate and channels or pick new ones. The decoding and writing happen in this tab, so nothing is uploaded and there is no queue.
Audio to WAV: why convert to the plainest format there is
WAV is the one audio format nearly every tool accepts. Editors, samplers, DJ software, transcription services, game engines, hardware recorders and decades-old broadcast playout systems all read it. They can because a PCM WAV has no codec: it is a 44-byte header followed by raw samples. Converting a compressed file to WAV does not restore what the MP3 or AAC encoder threw away. What it does give you is a copy that opens everywhere and can be edited and re-saved without losing any more quality.
Where WAV is the required format
- Speech recognition and transcription. Many APIs recommend or require 16 kHz mono 16-bit PCM, which is what their models were trained on.
- Telephony and IVR. Phone systems play prompts at 8 kHz mono. Asterisk and many hosted IVR platforms accept 8 kHz 16-bit WAV directly.
- Video and broadcast. Video editors and the EBU's Broadcast Wave format work at 48 kHz. Delivering audio at 48 kHz avoids a resample inside the edit.
- Samplers and game engines. Hardware samplers, trackers and older engines load WAV instantly because nothing needs decoding at runtime.
- Archiving an edit. If you will cut, fade or normalise a file several times, work in WAV and encode to MP3 once at the end.
What happens to your file
The browser's own decoder turns the file into 32-bit floating-point samples in memory. If you keep the source rate and channels, those samples are written straight out; nothing is resampled. Pick a new rate or Mono and the audio is rendered through an offline audio context at the target settings. That uses the browser's resampler, and Mono is the average of the channels. Each sample is then clipped to full scale and rounded to a 16-bit or 24-bit integer, without dither. The file is written with a standard PCM header and no metadata chunks, so ID3 tags and cover art from an MP3 are not carried over.
How big will the WAV be?
Size is simply sample rate × channels × bytes per sample × seconds. The WAV header stores sizes as 32-bit numbers, so a single file cannot exceed 4 GB, and the last column shows where that limit falls.
WAV size per minute and the 4 GB limit, by format
| Sample rate | Channels | Bits | Bit rate (kbps) | MB per minute | Hours to 4 GB |
|---|---|---|---|---|---|
| 8,000 | Mono | 16 | 128 | 0.92 | 74.6 |
| 16,000 | Mono | 16 | 256 | 1.83 | 37.3 |
| 22,050 | Mono | 16 | 352.8 | 2.52 | 27.1 |
| 32,000 | Mono | 16 | 512 | 3.66 | 18.6 |
| 44,100 | Mono | 16 | 705.6 | 5.05 | 13.5 |
| 44,100 | Stereo | 16 | 1,411.2 | 10.09 | 6.8 |
| 48,000 | Stereo | 16 | 1,536 | 10.99 | 6.2 |
| 44,100 | Stereo | 24 | 2,116.8 | 15.14 | 4.5 |
| 48,000 | Stereo | 24 | 2,304 | 16.48 | 4.1 |
| 96,000 | Stereo | 24 | 4,608 | 32.96 | 2.1 |
A 128 kbps MP3 is about 0.92 MB a minute, so the same song as CD-quality WAV is eleven times larger.
How to read the stats
- Source shows what the browser decoded: channels, sample rate, duration in seconds and the original file size. A 44,100 Hz reading for an MP3 is the rate it was encoded at, not a sign of quality.
- Output is the channel count and rate the WAV will have with your current settings.
- WAV size (16-bit) and WAV size (24-bit) are the exact sizes of the export for each bit depth, in megabytes of 1,048,576 bytes. The 24-bit file is always 50% larger.
Check the Output line before exporting. If it says 2 ch and you picked Mono, the selection has not been applied yet.
16-bit or 24-bit
16-bit is CD quality, with a theoretical dynamic range of about 96 dB. That is more than any MP3 or AAC source contains, so 16-bit loses nothing audible from a lossy file. Choose 24-bit when the source is a 24-bit FLAC or studio file, or when the WAV is going into heavy processing and you want the extra headroom below the noise floor.
Expert tips and pitfalls
- Loud masters may clip. Lossy decoders can produce peaks slightly above full scale on heavily limited music. Those samples are clipped at export. If you hear crackle on loud passages, lower the level first with a gain tool and convert again.
- Do not upsample for quality. Converting a 22 kHz file to 48 kHz adds no detail; do it only when a project demands a matching rate.
- Downsampling removes treble. A 16 kHz file holds nothing above 8 kHz. That is fine for speech, dull for music.
- Gapless albums. MP3 and AAC carry encoder padding. Some browsers trim it and some leave a few milliseconds of silence, which matters when joining tracks.
- Long files use memory. The whole WAV is built in the tab. Hour-long stereo recordings at 24 bits need several hundred megabytes of RAM.
- Protected files will not open. DRM-locked purchases cannot be decoded, and anything your browser cannot play is refused.
Going the other way, WAV to MP3 shrinks the file again. To change the rate or depth of an existing WAV, use the sample rate converter or the bit depth converter. The stereo to mono tool handles just the downmix.
Format-specific pages explain what each source codec keeps: MP3 to WAV, M4A to WAV and FLAC to WAV.
FAQ
Does converting an MP3 to WAV improve the sound?
No. The MP3 encoder already discarded detail and a WAV cannot bring it back. What you gain is a file that opens everywhere and can be edited repeatedly without losing any more, which is often reason enough to convert.
Which bit depth should I choose?
Sixteen bits unless you know you need more. It is what CDs, transcription services and most software expect. Choose 24 bits when the file is going to be processed further or when the source was 24-bit and you want a lossless copy.
Why is the WAV so much larger than the MP3?
Because WAV stores every sample uncompressed. Stereo at 44.1 kHz and 16 bits is about 1.4 megabits per second, roughly ten times a 128 kbps MP3. The size shown in the stats panel is exact for the settings you have chosen.
Can I convert several files at once?
Not on this page; it handles one file at a time so that the waveform and stats stay clear. Drop the next file after the first export finishes — the settings stay as you left them.
See also: all Audio & Video tools, Units, Images, Documents, Data & Code.
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