Convert AC3 to FLAC — Free Online Tool
Convert AC3 (Dolby Digital) audio files to FLAC, transforming lossy surround sound tracks into lossless compressed archives. This is especially useful for preserving 5.1 multichannel audio from DVD or Blu-ray sources in an open, high-fidelity format without further quality degradation.
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FFmpeg Command
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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 codec — it permanently discards audio information during encoding to achieve compression. When converting AC3 to FLAC, FFmpeg decodes the AC3 bitstream back to raw PCM audio, then re-encodes it using FLAC's lossless compression algorithm. The resulting FLAC file is a lossless representation of the decoded AC3 audio, meaning no additional quality is lost beyond what was already discarded when the AC3 was originally created. FLAC preserves the full multichannel layout (e.g., 5.1 surround) decoded from the AC3 source, storing every decoded sample bit-for-bit. The -compression_level parameter controls only how aggressively FLAC packs the data — it never affects audio fidelity, only file size and encoding speed.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. This command runs entirely in your browser via FFmpeg.wasm (WebAssembly) — no files leave your device. |
-i input.ac3
|
Specifies the input file, which is a raw AC3 (Dolby Digital) audio stream. FFmpeg will automatically detect the AC3 codec and prepare to decode it to PCM for re-encoding. |
-c:a flac
|
Sets the audio codec to FLAC (Free Lossless Audio Codec), instructing FFmpeg to encode the decoded AC3 PCM audio into FLAC's lossless compression format rather than any other codec. |
-compression_level 5
|
Sets the FLAC compression effort to level 5 (the standard default on a 0–8 scale). This controls how hard the encoder works to minimize file size — it never affects audio quality, only the tradeoff between encoding speed and output file size. |
output.flac
|
Defines the output filename with the .flac extension. FFmpeg uses this extension to confirm the target container and codec, writing the losslessly encoded audio from the decoded AC3 source into the FLAC file. |
Common Use Cases
- Archiving the Dolby Digital audio track ripped from a DVD to a lossless format for long-term preservation without further lossy re-encoding
- Preparing a 5.1 surround sound AC3 track extracted from a broadcast recording for editing in a DAW that supports FLAC but not AC3
- Converting AC3 audio from a Blu-ray remux into FLAC so it can be embedded in an MKV container alongside a video track for media server playback
- Migrating a library of AC3 files to FLAC for compatibility with music players and NAS-based media systems like Plex or Jellyfin that handle FLAC natively
- Decoding an AC3 track to FLAC before applying mastering or restoration processing, ensuring no additional lossy compression is introduced in the audio chain
- Storing decoded AC3 surround audio as FLAC for use as a lossless reference when comparing different audio codec encodes
Frequently Asked Questions
No. AC3 is a lossy codec, meaning audio data was permanently discarded when the AC3 file was originally encoded. Converting to FLAC captures the decoded output of that lossy file losslessly — so the FLAC will be a perfect, bit-exact copy of what comes out of the AC3 decoder, but it cannot recover the information that was lost during the original AC3 encoding. Think of it as losslessly preserving a lossy source: no further quality is lost, but none is regained either.
Yes. FLAC fully supports multichannel audio including 5.1 and 7.1 layouts. When FFmpeg decodes an AC3 5.1 track, it outputs six discrete audio channels (front left, front right, center, LFE, surround left, surround right), and FLAC encodes all of them losslessly. The channel layout metadata is preserved in the FLAC file, so compliant players will correctly map the surround channels on playback.
AC3 achieves its small file size through lossy compression, discarding audio data the encoder deems perceptually irrelevant. FLAC, even though it compresses efficiently, stores every decoded audio sample losslessly — so it is inherently representing far more data than the AC3 was. A typical AC3 file at 192–448 kbps might expand to a FLAC file several times larger, because FLAC is preserving the full uncompressed PCM resolution of the decoded audio.
The -compression_level flag in FLAC controls the encoding effort used to compress the audio data, on a scale from 0 (fastest, least compression) to 8 (slowest, smallest file). It has absolutely no effect on audio quality — all levels produce bit-identical decoded audio. Level 5 is the standard default that balances speed and file size well for most use cases. For AC3 source files, the differences in output file size between levels are usually modest, so the default is a sensible choice.
Replace the value after -compression_level in the command. For example, to use maximum compression (smallest file, slowest encode), run: ffmpeg -i input.ac3 -c:a flac -compression_level 8 output.flac. To prioritize encoding speed at the cost of slightly larger files, use level 0 or 1. Remember that regardless of the level chosen, the decoded audio output is always identical — only encoding time and file size are affected.
Yes. On Linux or macOS you can use a shell loop: for f in *.ac3; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.ac3}.flac"; done. On Windows Command Prompt, use: for %f in (*.ac3) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". This processes each AC3 file in the current directory and outputs a corresponding FLAC file with the same base name.
Technical Notes
AC3 supports up to 5.1 channels at bitrates ranging from 32 kbps to 640 kbps, and the decoded PCM output is always 16-bit at either 44.1 kHz or 48 kHz (48 kHz being standard for DVD and broadcast sources). FLAC will losslessly encode whichever sample rate and bit depth the AC3 decoder outputs — typically 16-bit/48 kHz for broadcast-origin files. Note that while FLAC supports up to 32-bit integer PCM, the AC3 decoder outputs at most 16-bit or 24-bit depending on the FFmpeg build, so the FLAC bit depth will reflect that. Metadata handling is limited: AC3 files carry minimal metadata (primarily codec configuration and bitstream headers), so the resulting FLAC file will generally have sparse tags. You may want to add FLAC's Vorbis comment metadata (artist, title, album, etc.) using a tag editor after conversion. FLAC also natively supports ReplayGain tags, which can be applied post-conversion for consistent volume normalization across a library. There are no subtitle, chapter, or cover art streams in AC3, so the conversion is a straightforward single-stream audio transcode.