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AAC to WAV converter

Convert AAC audio to lossless WAV with configurable sample rate and bit depth.

Free & unlimited
All processing happens in your browser — files never leave your device. The audio engine loads once from a CDN.

A bare .aac file is a raw ADTS stream with no container around it — no chapter marks, no reliable tags, and no support in a surprising number of editors. Converting it to WAV gives you uncompressed PCM that every piece of audio software opens without a codec. This tool decodes the AAC and writes PCM in your own browser using ffmpeg compiled to WebAssembly. Four presets set sample rate and bit depth together: Studio at 96 kHz 32-bit float, Hi-fi at 48 kHz 24-bit, CD at 44.1 kHz 16-bit, and Voice at 22.05 kHz 16-bit for speech. Batches convert one after another and come back as a ZIP built in the browser.

Key facts about AAC to WAV converter

Key facts about AAC to WAV converter
What .aac usually isA raw ADTS AAC stream with no container — distinct from the AAC inside an .m4a or an .mp4
What changesThe AAC bitstream is decoded to PCM. Nothing is recovered; nothing further is lost
Studio preset96 kHz, 32-bit float — maximum headroom for a mixing chain
Hi-fi preset48 kHz, 24-bit — the professional and broadcast standard rate
CD preset44.1 kHz, 16-bit — the standard lossless target and the sensible default
Voice preset22.05 kHz, 16-bit — a quarter of the size of CD rate, ample for speech
Typical size growthA 5 MB AAC becomes roughly 50 MB at CD preset; the Studio preset is several times larger again
Ratio shownThe result panel reports the output-to-input size ratio so the growth is visible before you download
TagsRaw AAC carries little metadata and WAV has no reliable tagging, so treat the output as untagged
Size cap200 MB per input file

What happens to your file

The decode and PCM write run on ffmpeg compiled to WebAssembly in a Web Worker inside this tab. Your AAC files are written into that worker's in-memory filesystem and the WAVs come back as blobs; the batch ZIP is built in the browser. Nothing is uploaded and nothing is stored. The single network request is the first-time download of the ffmpeg core, about 32 MB of fixed public code from the jsDelivr CDN, cached by your browser and shared across every audio tool here — it contains no part of your files.

About this tool

  1. 1

    Add your AAC files

    Drop one or several .aac files, each up to 200 MB. Raw AAC streams from stream rippers and some recorders land here.

  2. 2

    Pick a preset

    CD for general use, Hi-fi when the file is going into a professional or video project at 48 kHz, Voice for speech, Studio only when a mixing chain genuinely wants 32-bit float.

  3. 3

    Check the projected size

    The interface shows the ratio between input and output. At the Studio preset the growth is dramatic, so pick it deliberately rather than by default.

  4. 4

    Convert

    Files process one at a time with per-file progress. The engine loads once and is reused for the rest of the batch.

  5. 5

    Play it back

    The result can be auditioned in the page before you download, which is a quick check that the raw stream decoded cleanly.

  6. 6

    Download

    Take files individually or grab the whole batch as a ZIP.

Specs & compatibility
InputAAC (.aac, audio/aac), up to 200 MB per file
OutputUncompressed PCM WAV
Sample rates22.05, 44.1, 48 or 96 kHz, set by the chosen preset
Bit depths16-bit and 24-bit integer, 32-bit float
Output size ruleSample rate x bit depth x channels x seconds / 8 — about 10 MB per stereo minute at CD preset
Batch outputIndividual downloads or one client-side ZIP
EngineSingle-threaded ffmpeg.wasm 0.12.10, ~32 MB core cached after first use
BrowsersChrome, Edge, Firefox and Safari 15+
  • Use the CD preset unless you have a specific reason not to. 44.1 kHz 16-bit already exceeds anything the AAC source contained.
  • The Voice preset is genuinely useful for speech: at 22.05 kHz 16-bit a file is a quarter the size of CD rate and sounds identical for dialogue.
  • The Studio preset upsamples. Interpolating to 96 kHz produces more samples, not more information — pick it because a project requires it, not for quality.
  • If you are archiving rather than editing, FLAC is a better target than WAV: lossless, roughly half the size, and properly taggable.
  • A bare .aac file that some software refuses to open often works once it is remuxed into an .m4a — the general audio converter can do that too.
  • WAV has no dependable tagging, so keep track of filenames before converting a batch of similarly named files.
  • If the AAC is quiet, follow up with the audio normalizer rather than trying to fix level during conversion.
  • Four quality presets from voice to studio
  • 16-bit, 24-bit and 32-bit float output
  • Sample rates from 22.05 kHz to 96 kHz
  • Size ratio reported for each converted file
  • In-page playback of the result
  • Batch conversion with per-file progress
  • ZIP download for multiple files
  • Opening a raw AAC stream in editing software that refuses the format
  • Preparing AAC audio for a game engine or a tool that only reads PCM
  • Getting a 48 kHz 24-bit WAV out of an AAC recording for a video edit
  • Feeding an AAC file into a transcription or analysis pipeline that expects WAV
  • Converting recorded stream audio into an editable uncompressed format
  • Making a voice-rate WAV from a speech recording for a small embedded player
The codec is the same; the wrapper is not. A bare .aac file is a raw ADTS stream — just AAC frames, one after another, with no container structure. An .m4a is an MPEG-4 container holding AAC (or sometimes Apple Lossless) along with metadata, chapter marks and seek information. The container is why .m4a files tag properly and seek reliably, and why raw .aac sometimes fails to open in software that expects a container. Both decode to the same PCM here.
No. AAC is lossy, so information was discarded permanently when the file was made, and decoding it to PCM cannot restore any of it. WAV stores the decoded result exactly, without adding further loss — that is the benefit. What you gain is universal compatibility and a guarantee that later editing steps do not introduce another lossy generation. What you pay is size: expect the file to be roughly ten times larger at the CD preset.
CD, at 44.1 kHz 16-bit, for almost everything — it already carries more dynamic range and bandwidth than the AAC source contained. Hi-fi at 48 kHz 24-bit when the audio goes alongside video, since 48 kHz is the video standard and matching it avoids a resample later. Voice at 22.05 kHz 16-bit for speech, which cuts the size to a quarter with no audible cost. Studio at 96 kHz 32-bit float only when a mixing chain specifically requires float headroom.
It follows arithmetic rather than content: sample rate times bit depth times channels times seconds, divided by eight. At the CD preset that is about 10 MB per stereo minute, so a 5 MB AAC of about thirty-five minutes becomes something near 350 MB. The Hi-fi preset is roughly 1.6 times that and the Studio preset several times more again. The result panel shows the ratio for each file so you can see the growth before committing to a download.
Yes. Drop as many .aac files as you like and they convert in sequence with per-file progress, using the same preset. You can download each one individually or take the whole set as a ZIP, which is assembled inside your browser from the converted blobs rather than on a server. Because everything is held in memory, a very large batch of WAVs is limited by your device rather than by a quota — WAV files add up quickly, so convert in reasonable groups.
No. The decode and PCM write are performed by ffmpeg compiled to WebAssembly, running in a Web Worker on your own machine. Your AAC files go into that worker's in-memory filesystem and the WAVs come back as blobs in your tab; the ZIP is built in the browser too. The only network request is the one-time fetch of the ffmpeg core, about 32 MB of fixed public code from the jsDelivr CDN that is cached and reused across every audio tool here.

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