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M4A to WAV converter

Convert M4A audio to uncompressed WAV with bit depth and sample rate control.

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

WAV is what editing software, DAWs, game engines and broadcast systems want: uncompressed PCM with no decoding step and no codec dependency. M4A is what Apple devices produce. This converter decodes the AAC (or Apple Lossless) inside your M4A and writes straight PCM WAV, running on ffmpeg in your browser. You choose the bit depth — 16-bit for standard work, 24-bit for a professional chain, 32-bit float when the file is heading into a mix and you want headroom above full scale — and the sample rate, from 22.05 kHz up to 96 kHz. An optional normalize pass lifts the peak to 0 dB. Batches convert in sequence and download as a ZIP.

Key facts about M4A to WAV converter

Key facts about M4A to WAV converter
What changesThe AAC bitstream is decoded to PCM samples; nothing is recovered, but nothing further is lost either
Bit depths16-bit (standard), 24-bit (professional) and 32-bit float (mixing headroom)
Sample rates22.05 kHz, 44.1 kHz, 48 kHz and 96 kHz
Typical size growthA 5 MB M4A becomes roughly 50 MB of 16-bit 44.1 kHz stereo WAV — about ten times larger
24-bit costFifty percent larger again than 16-bit, for a dynamic range the AAC source never contained
UpsamplingChoosing 96 kHz from a 44.1 kHz source interpolates samples; it does not add detail that was never encoded
Optional normalizeRaises the peak to 0 dB during conversion
MetadataWAV has no standard tagging scheme worth relying on, so treat the output as untagged audio
Compatibility gainedEvery DAW, editor, game engine and broadcast system opens WAV with no codec required
Size cap200 MB per input file

What happens to your file

The decode and PCM write happen in ffmpeg compiled to WebAssembly, running in a Web Worker inside this tab. Your M4A files go into that worker's in-memory filesystem and the WAVs come back as blobs; the batch ZIP is assembled in the browser. Nothing is uploaded and nothing is retained. The only network request is the first-time download of the ffmpeg core, about 32 MB of fixed public code from the jsDelivr CDN, cached afterwards and reused by every audio tool here. It carries no part of your audio.

About this tool

  1. 1

    Add your M4A files

    Drop one file or several. Watch the size: an M4A near the 200 MB cap becomes an enormous WAV.

  2. 2

    Pick a bit depth

    16-bit unless something downstream needs more. 24-bit for a professional edit chain. 32-bit float when the file goes into a mix where intermediate stages might exceed 0 dBFS.

  3. 3

    Pick a sample rate

    Match the source or match the project. 44.1 kHz for music, 48 kHz for anything going with video. Do not upsample hoping for better quality.

  4. 4

    Decide on normalize

    Turning it on raises the peak to 0 dB, which is handy for a quiet voice memo and unhelpful for a file that will be mixed with others.

  5. 5

    Convert

    Files process one at a time with progress. Output sizes are reported so you can see what you are about to download.

  6. 6

    Download

    Take the WAVs individually or as a ZIP, then import them into whatever needed uncompressed PCM.

Specs & compatibility
InputM4A (.m4a, audio/mp4, audio/x-m4a), up to 200 MB per file
OutputUncompressed PCM WAV
Bit depths16-bit integer, 24-bit integer, 32-bit float
Sample rates22.05, 44.1, 48 and 96 kHz
Output size ruleSample rate x bit depth x channels x seconds, divided by eight — roughly 10 MB per stereo minute at 16-bit 44.1 kHz
Quality presetsStudio 96 kHz 32-bit float, Hi-fi 48 kHz 24-bit, CD 44.1 kHz 16-bit, Voice 22.05 kHz 16-bit
EngineSingle-threaded ffmpeg.wasm 0.12.10
BrowsersChrome, Edge, Firefox and Safari 15+
  • Converting to WAV does not improve the audio. The AAC inside your M4A already discarded information permanently; WAV simply stores what is left without further loss.
  • 16-bit at 44.1 kHz is the right default. 24-bit stores a dynamic range the AAC source never had, at fifty percent more size.
  • Use 32-bit float only when the WAV is going into a mix. It lets intermediate stages exceed 0 dBFS without clipping, which is its actual purpose.
  • 48 kHz if the audio pairs with video — that is the video standard, and matching it avoids a resample in the edit.
  • WAV files have no reliable tagging, so if the tracks came from a library, note the filenames before you convert or you will lose track of what is what.
  • For archiving rather than editing, FLAC is the better target: lossless like WAV but roughly half the size. The general audio converter does that.
  • Voice memos convert perfectly well at 22.05 kHz 16-bit, which is a quarter of the size of a CD-rate stereo file.
  • 16-bit, 24-bit and 32-bit float output
  • Sample rates from 22.05 kHz to 96 kHz
  • Four quality presets from voice to studio
  • Optional peak normalization to 0 dB
  • Handles both AAC and Apple Lossless inside .m4a
  • Batch conversion with per-file progress
  • ZIP download for multiple files
  • Importing iPhone voice memos into a DAW that will not read M4A
  • Preparing audio for a game engine that requires uncompressed PCM
  • Getting an interview recording into editing software with no codec dependency
  • Feeding an M4A into a transcription or analysis tool that only accepts WAV
  • Making a broadcast-standard 48 kHz 24-bit file from a phone recording
  • Converting an Apple Lossless file into WAV for a mastering chain
No. This is the most common misunderstanding about lossless containers. The AAC inside an M4A has already discarded audio information permanently, and decoding it to PCM cannot bring any of it back — WAV simply stores the decoded result exactly as it is, without adding further loss. What you gain is compatibility and the guarantee that no additional damage happens in later editing steps. What you pay is file size: roughly ten times the original.
16-bit unless something specific demands more. It is the CD standard, it is what most software expects, and it already exceeds the dynamic range your AAC source contained. Choose 24-bit when a professional chain requires it, for instance a mastering handoff. Choose 32-bit float when the WAV is going into a mix: float files can hold values above 0 dBFS without clipping, so intermediate processing stages have headroom, which is the one genuinely functional reason to use it.
Because it is uncompressed. A WAV stores every sample at full precision with no encoding at all, so its size follows a simple formula: sample rate times bit depth times channels times duration. At 16-bit 44.1 kHz stereo that is about 10 MB per minute regardless of the material, where an AAC file at 192 kbps is closer to 1.4 MB per minute. A ten-times increase is the normal outcome, and 24-bit or 96 kHz multiplies it further.
Only if something downstream requires it. Interpolating a 44.1 kHz recording up to 96 kHz produces more samples but no more information — the frequencies above 22 kHz that a 96 kHz file could theoretically hold simply were not in the source and cannot be invented. Upsampling is occasionally required to match a project's rate, and that is a legitimate reason. Doing it in the hope of better sound is not; it makes the file more than twice as large for nothing.
Effectively no. WAV has no widely respected tagging standard — there are RIFF INFO chunks and ID3-in-WAV conventions, but support for them is inconsistent enough that no sensible workflow depends on them. Treat the output as untagged audio and rely on filenames. If keeping metadata matters, convert to FLAC instead using the general audio converter: it is equally lossless, roughly half the size, and has a proper tagging system that every player understands.
No. The decode and PCM write are done by ffmpeg compiled to WebAssembly, running in a Web Worker on your own device. Your M4A goes into that worker's in-memory filesystem and the WAV comes back as a blob in your tab. Nothing crosses the network. The one request the tool makes is the first-time fetch of the ffmpeg core — about 32 MB of fixed public code from the jsDelivr CDN — which is the same for every visitor and cached afterwards.

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