Convert AAC to WAV
Decode a raw .aac stream into a 16-bit PCM WAV with the same sample rate and channels, so any editor can open it.
Decode a raw .aac stream into a 16-bit PCM WAV with the same sample rate and channels, so any editor can open it.
A file ending in .aac is a bare AAC stream, stored in a framing format called ADTS without the MP4 wrapper that an M4A file has. That bare format is convenient for streaming and for pulling audio out of video without re-encoding, but it is awkward in practice. Many editors won't import it, and because ADTS has no index of where each second begins, some players show the wrong duration or struggle to seek.
Converting to WAV removes all of that. You get a 16-bit PCM WAV at the same sample rate and with the same channels as the AAC stream, ready for any audio program. The settings are fixed, so there is nothing to adjust before you convert.
AAC is lossy, and decoding it doesn't reverse that. The WAV contains the same sound the AAC produces on playback, no better and no worse, just stored without compression. What decides the WAV's size is not the AAC bitrate but the sample rate and number of channels, both of which are kept:
| AAC stream | WAV per minute | Notes |
|---|---|---|
| Mono, 44.1 kHz | about 5 MB | Common for voice recordings |
| Stereo, 44.1 kHz | about 10.1 MB | Typical for music |
| Stereo, 48 kHz | about 11 MB | Typical for audio taken from video |
| 5.1 surround, 48 kHz | about 33 MB | All six channels are written |
Editing clip audio. After extracting a soundtrack or interview from a video, WAV is the format a DAW or audio editor imports most reliably, so you can clean up noise, cut and mix without format errors.
Speech recognition and captioning. Transcription software often prefers PCM input, and a WAV avoids decoder problems with raw streams.
Sampling. A line or sound grabbed from a stream recording can be loaded into a sampler or drum machine that only reads WAV.
Accurate length and seeking. If a player jumps around or reports a strange running time for your AAC file, the WAV will report its length exactly.
Surround AAC, common in audio pulled from films and TV, stays 5.1 in WAV. Some editors and players expect stereo. Our MP3 output folds surround down to stereo, so AAC to MP3 is an option when two channels are all you need.
A file named .aac is sometimes something else, such as AC3 or MP2 audio from a TV capture, and truncated stream recordings may be damaged. FFmpeg can usually read the real format and skip small glitches, but a badly broken file may not convert.
Nothing is resampled. If a program needs 44.1 kHz and your stream was 48 kHz, that program will have to convert the rate itself.
If your file ends in .m4a, use M4A to WAV. The codec is the same, and the result is identical in format. Raw AAC rarely carries tags, so there is little metadata to lose either way. Other outputs for these files are on the AAC converter.
| AAC | WAV | |
|---|---|---|
| Full name | Advanced Audio Coding (ADTS stream) | Waveform Audio File Format |
| Compression | Lossy | Uncompressed (PCM) |
| Developed by | MPEG (Fraunhofer, Dolby, Sony, Nokia and others) | Microsoft and IBM |
| Introduced | 1997 | 1991 |
| File extension | .aac | .wav |
| Typically used for | Streaming, broadcast, audio from recorders and apps | Recording, editing, mastering, broadcast |
| Plays on | iPhone, Android, modern browsers and most media players | All desktop systems and audio editors; most phones and browsers |
| Our output | AAC-LC, 192 kbps, raw ADTS stream | 16-bit PCM, original sample rate and channels |
Both usually contain the same AAC audio. A .aac file is a bare stream in ADTS framing, while .m4a wraps AAC in an MP4 container that can hold tags and an index for seeking. Raw AAC is common when audio is copied out of video or recorded from streams, and many editors handle it poorly.
No. AAC is lossy, so decoding to WAV can't bring back what the encoder removed. The WAV is an uncompressed 16-bit PCM copy of exactly what the AAC sounds like. It is useful because it opens reliably in editors, samplers and transcription software, and it reports its duration accurately.
All six channels. WAV output keeps the channel layout of the source, so surround AAC becomes a 6-channel WAV at about 33 MB per minute at 48 kHz. If your editor or player expects stereo, convert to MP3 instead, which folds surround down to two channels.
The upload limit is 50 MB, which is about 35 minutes of 192 kbps AAC or around 70 minutes at 96 kbps. The WAV you download isn't limited, but it will be large: about 10.1 MB per minute of 44.1 kHz stereo, so a 35-minute file becomes roughly 350 MB.
Sometimes a file named .aac actually contains a different format, such as AC3 or MP2 from a TV capture, or a stream recording was cut off and damaged. FFmpeg often detects the real format and skips small errors, but badly corrupted files may not convert. Files over 50 MB are also rejected.
Need detailed information or want to set a different default format? Visit our specific converter pages: