AAC to WAV converter

A QC meter cannot read a stream, only PCM: a raw AAC file is an elementary stream produced by an encoder, and measurement tools need decoded samples to produce numbers. This route is about inspection, not about making the file playable for a listener.

WAV stores raw PCM, so the file is much larger than the compressed source.

Drop files here or Choose files AAC, up to 60 MB

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The file is a stream, and the meter wants samples

Illustrative: a streaming pipeline left behind a raw AAC audio track from an earlier encode, and a QC engineer needs loudness numbers and a trim before the archive copy is written. The QC tool refuses the file because it is an elementary stream. Decoding is the step that turns an unreadable stream into measurable PCM.

One file per run, up to 60 MiB and 600 seconds of audio.

Drag the raw AAC file onto this page; the local FFmpeg worker loads on first use
Choose the sample depth your measurement tool expects — 16-bit by default
Convert, then read the reported codec, duration, sample rate and channel count
Run the loudness meter and the trim on the decoded WAV

Diagnosing the stream before trusting the numbers

A stream can be intact, shortened, or header-damaged, and the three behave differently after decoding. Duration is the diagnostic that separates them.

Condition	What the decode shows	What to do
Intact stream	Duration matches the source	Measure and trim normally
Shortened stream	Duration shorter than expected, reported as failure	Find the complete stream before measuring
Damaged ADTS framing	Decode fails or produces a short file	Recover the stream from the pipeline, not from this file

Why the decoded duration is the only trustworthy signal here

A container carries explicit length metadata; an elementary stream does not. With ADTS, each frame declares its own size, so a truncated or mis-framed stream still decodes — it just decodes to less audio than you expected. Compare the reported duration against the length you know the track should have before you feed the WAV to a meter.

Inspection questions

Why does a player struggle with a raw .aac file?

Raw AAC is an elementary stream, typically ADTS-framed, without a container. Many players expect AAC inside MP4, so playback is inconsistent. Decoding to WAV is the reliable way to inspect it.

Can loudness numbers be trusted on decoded audio?

They describe the audio that survived the encode. AAC discarded information at encode time, so a loudness fix is applied to what exists now and must be re-encoded, not patched into the original stream.

What if the stream has damaged headers?

A stream with broken ADTS framing can decode shorter than expected. The engine compares decoded duration with the source and reports a failure rather than handing over a truncated WAV.

Does decoding restore discarded detail?

No. Decoding reconstructs what survived; it cannot recover audio information the encoder discarded.

Is the stream uploaded?

No. Decoding runs inside this page through a self-hosted FFmpeg worker; the first run loads the WASM core from this site over http or https.

Editing a decoded stream means re-encoding

There is no way to patch a fix into the original elementary stream. Once you trim or adjust levels, the result is new audio and needs a new encode.

Decode to WAV, apply the loudness fix and trim, then encode from the edited WAV
Encode for the consumer: raw AAC for a muxer, M4A for devices, MP3 for a legacy host
Keep the decoded WAV as a working file, not as the archive copy
Record in the QC log that the measurement was taken on decoded audio

What this route is not for

It is not a way to make a listener-friendly file, and it is not a repair tool.

For a deliverable that plays everywhere: encode from the source master, not from a leftover stream
For container-wrapped phone recordings: use M4A to WAV
For game assets stored as Ogg: use Ogg to WAV

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