Convert AC3 to WAV — Free Online Tool
Convert AC3 (Dolby Digital) audio files to uncompressed WAV format using PCM encoding — directly in your browser. This tool decodes the lossy AC3 bitstream and outputs a 16-bit PCM WAV file, making it ideal for professional audio workflows that require uncompressed source material.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Settings
Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
AC3 is a lossy compressed audio format that encodes audio — often multichannel surround sound — using psychoacoustic compression to reduce file size. During conversion, FFmpeg decodes the AC3 bitstream entirely, reversing the compression to reconstruct PCM audio samples, then writes them into a WAV container using the pcm_s16le codec (signed 16-bit little-endian PCM). It is important to understand that this is a lossy-to-lossless container conversion: the WAV output will be uncompressed, but any audio quality lost during the original AC3 encoding cannot be recovered. If the source AC3 is multichannel (e.g., 5.1 surround), the channel layout will be preserved in the output WAV file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, FFmpeg runs as a WebAssembly module (FFmpeg.wasm) entirely within your browser — no files leave your device. The same command runs identically on a local FFmpeg installation for files over 1 GB. |
-i input.ac3
|
Specifies the input file — an AC3 (Dolby Digital) audio file. FFmpeg reads and decodes the AC3 bitstream, handling all standard channel configurations from mono through 5.1 surround at any supported Dolby Digital bitrate. |
-c:a pcm_s16le
|
Sets the audio codec for the output to PCM signed 16-bit little-endian, which is the standard uncompressed audio encoding for WAV files. This is the format used by audio CDs and is universally supported across DAWs, editing software, and consumer media players. |
output.wav
|
Defines the output filename with a .wav extension. FFmpeg uses this extension to automatically select the RIFF WAV container format, which will wrap the decoded pcm_s16le audio stream into a file compatible with virtually all audio software. |
Common Use Cases
- Extracting dialogue or sound effects from a DVD or Blu-ray AC3 audio track for use in a video editing timeline that requires uncompressed PCM audio
- Importing broadcast television audio delivered in AC3 format into a Digital Audio Workstation (DAW) like Pro Tools or Logic Pro, which natively handles WAV/PCM
- Archiving AC3 audio from legacy media to uncompressed WAV for long-term storage in a format with maximum software compatibility
- Preparing AC3 surround sound audio for re-encoding into a different codec (e.g., AAC or Opus) by first converting to an uncompressed intermediate to avoid generation loss
- Converting AC3 audio tracks from video game cutscenes or DVD extras for use in multimedia projects or fan restoration efforts
- Testing and quality-comparing an AC3-encoded track against its original source by decoding to PCM for waveform analysis in an audio editor
Frequently Asked Questions
No. While WAV with PCM encoding is a lossless, uncompressed format, converting from AC3 does not recover quality lost during the original Dolby Digital compression. The resulting WAV file is an exact uncompressed representation of what the AC3 decoder outputs — which already reflects any artifacts introduced when the AC3 was originally encoded. Think of it as unpackaging the compressed audio, not restoring it.
FFmpeg preserves the full multichannel layout from the AC3 source by default. If your AC3 file contains 5.1 audio (front left, front right, center, LFE, surround left, surround right), the output WAV will contain all six channels encoded as 16-bit PCM. Most professional DAWs and audio editors can handle multichannel WAV files, though some consumer software may only play the front stereo pair.
AC3 is a lossy compressed format that typically achieves compression ratios of 10:1 or more compared to uncompressed audio. A 5.1 AC3 file encoded at 192 kbps will expand dramatically when decoded to 16-bit PCM at 48 kHz — the same audio uncompressed at CD quality requires roughly 4.6 Mbps for a stereo signal, and over 13 Mbps for 5.1. This size increase is completely expected and not a sign of any problem with the conversion.
Replace pcm_s16le with pcm_s24le in the command: ffmpeg -i input.ac3 -c:a pcm_s24le output.wav. This outputs 24-bit signed little-endian PCM, which is preferred in professional audio workflows. Other valid options for WAV include pcm_s32le (32-bit integer) and pcm_f32le (32-bit floating point). Note that since AC3 is a lossy source, the additional bit depth won't recover lost information but can be useful to match a project's target specification.
Yes, on the command line you can use a shell loop. On Linux or macOS: for f in *.ac3; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.ac3}.wav"; done. On Windows Command Prompt: for %f in (*.ac3) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". The browser-based tool processes one file at a time, but the displayed FFmpeg command is exactly what you'd use locally for batch scripting.
AC3 files carry limited metadata compared to container formats like MKV or MP4. WAV also has restricted metadata support — it uses a basic INFO chunk that can hold fields like artist and title, but this is inconsistently supported across software. FFmpeg will attempt to map any available metadata tags from the AC3 source into the WAV output, but do not expect rich tagging to transfer reliably. For metadata-heavy workflows, consider MP3 or FLAC as output formats instead.
Technical Notes
The FFmpeg command uses pcm_s16le as the output codec, which is the WAV format's most universally compatible audio encoding — signed 16-bit integers in little-endian byte order, matching the CD audio standard. The input AC3 stream is first decoded by FFmpeg's built-in AC3 decoder, which handles all standard Dolby Digital bitrates from 96 kbps up to 640 kbps, as well as mono, stereo, and full 5.1/6-channel configurations. Since WAV does not support subtitles, chapters, or multiple audio tracks, this conversion is purely a single-stream audio operation. One practical limitation: WAV files using standard PCM encoding have a 4 GB file size ceiling due to the 32-bit chunk size field in the RIFF container header — a constraint relevant for very long multichannel recordings. The sample rate from the AC3 source (typically 48 kHz for broadcast/DVD content) is passed through unchanged by default, so the output WAV will be 48 kHz PCM rather than 44.1 kHz (CD standard) unless you explicitly add a -ar 44100 flag to resample.