Convert AC3 to AAC — Free Online Tool
Convert AC3 (Dolby Digital) audio files to AAC format, transcoding the multi-channel surround sound stream into a widely compatible lossy format optimized for streaming, mobile playback, and web delivery. This is especially useful when you need to repurpose DVD or broadcast audio content for Apple devices, web players, or mobile apps that don't support the AC3 codec natively.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 and AAC are both lossy audio codecs, so this conversion is a full transcode — the audio is fully decoded from the Dolby Digital AC3 bitstream back to uncompressed PCM, then re-encoded using AAC compression. There is no lossless path between these two formats. The AC3 source may contain 5.1 surround sound channels (front left, front right, center, LFE, surround left, surround right), and FFmpeg's built-in AAC encoder will preserve the channel layout in the output AAC file. Because both formats are lossy, you should use the highest practical bitrate to minimize the cumulative quality degradation introduced by this double-encoding step.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which is the underlying engine performing this AC3-to-AAC transcode. In the browser, this runs via FFmpeg.wasm — the same logic compiled to WebAssembly. |
-i input.ac3
|
Specifies the input file — a raw Dolby Digital AC3 audio file. FFmpeg reads and demuxes the AC3 bitstream, which may contain stereo or multi-channel (e.g., 5.1) audio, and decodes it to PCM for re-encoding. |
-c:a aac
|
Selects FFmpeg's built-in AAC encoder to re-encode the decoded audio. This produces AAC-LC output, which is the most broadly compatible AAC profile and is natively supported by Apple devices, Android, and all major web browsers. |
-b:a 128k
|
Sets the AAC output bitrate to 128 kilobits per second. For stereo content this is generally acceptable, but for 5.1 surround sound carried over from AC3, consider raising this to 192k or 256k to better preserve the multi-channel audio detail. |
output.aac
|
Defines the output filename with the .aac extension, telling FFmpeg to write a raw AAC audio stream. This file will play natively on Apple devices and can be used as an audio source in web and mobile applications. |
Common Use Cases
- Extracting Dolby Digital audio from a DVD rip or broadcast recording and converting it to AAC for playback on an iPhone, iPad, or Apple TV, which don't support AC3 natively in all app contexts.
- Preparing surround sound audio tracks for upload to streaming platforms or web video players that require AAC audio in an MP4 or HLS container.
- Converting AC3 audio from a Blu-ray or DVD source to AAC so it can be used as a soundtrack in iMovie, Final Cut Pro, or other Apple editing tools.
- Repurposing broadcast television audio (commonly encoded as AC3) into AAC for use in a podcast, YouTube video, or social media clip.
- Converting a Dolby Digital AC3 file to AAC to reduce file size for distribution over mobile networks, where AAC is more universally supported at lower bitrates.
- Archiving or remuxing video files into MP4 containers where the video stream is copied but the AC3 audio must be transcoded to AAC for broad device compatibility.
Frequently Asked Questions
Yes, some quality loss is unavoidable. AC3 is already a lossy format, and AAC is also lossy, so transcoding between them means the audio is decompressed and then re-compressed — a process sometimes called generation loss. To minimize this, use a higher output bitrate (e.g., 192k or 256k) rather than the default 128k. The audible difference at 192k AAC versus the source AC3 is usually subtle for most listeners, but the conversion is not mathematically transparent.
Yes. AAC supports multi-channel audio including 5.1 surround sound, and FFmpeg will preserve the channel layout from the AC3 source when encoding to AAC. If your AC3 file contains a 5.1 channel configuration, the output AAC file will also be 5.1. However, compatibility with 5.1 AAC depends on the playback device — most modern Apple devices and browsers handle it correctly, but some older or simpler players may downmix to stereo.
AC3 (Dolby Digital) decoding requires a license, so Apple's iOS and macOS devices only support AC3 playback in limited contexts (e.g., pass-through over HDMI or in specific apps). Web browsers do not support AC3 at all via the HTML5 Audio or Video API. AAC, by contrast, is natively supported on all Apple devices, Android, and all major browsers, making it the practical choice for broad compatibility.
Replace the value after '-b:a' in the command to set a different bitrate. For example, to encode at 256 kbps instead of 128 kbps, use: ffmpeg -i input.ac3 -c:a aac -b:a 256k output.aac. For 5.1 surround content from an AC3 source, a bitrate of 192k–320k is recommended to preserve perceptible surround detail, as 128k may introduce audible artifacts on a multi-channel mix.
The output file size depends heavily on the bitrate you choose. A typical AC3 file for 5.1 audio is encoded at 384k or 448k, so converting to AAC at the default 128k will produce a significantly smaller file — roughly one-third the size. If you match the bitrate (e.g., encoding AAC at 384k), the file sizes will be similar, though AAC tends to achieve slightly better quality-per-bit than AC3, so you can often use a lower AAC bitrate and retain comparable perceived quality.
Yes. On Linux or macOS, you can loop over all AC3 files in a directory with a single shell command: for f in *.ac3; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.ac3}.aac"; done. On Windows Command Prompt, use: for %f in (*.ac3) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". The browser-based tool processes one file at a time, so the FFmpeg command is the best option for batch processing large collections.
Technical Notes
The FFmpeg built-in AAC encoder (codec identifier 'aac') is used here, which produces standards-compliant AAC-LC (Low Complexity) output compatible with virtually all modern devices. An alternative is 'libfdk_aac', which is widely regarded as producing higher quality output at the same bitrate, but it requires a separately compiled version of FFmpeg with libfdk_aac enabled due to licensing restrictions — it is not available in most pre-built FFmpeg binaries or in FFmpeg.wasm. The AC3 format can carry metadata such as dialogue normalization (dialnorm) and dynamic range control (DRC) values embedded in the Dolby Digital bitstream; these are bitstream-level parameters specific to AC3 and will not be transferred to the AAC output, since AAC does not have equivalent fields. Standard metadata tags (title, artist, etc.) are also typically absent from raw AC3 files. If your AC3 source contains more than 5.1 channels (uncommon but possible in some broadcast contexts), the FFmpeg AAC encoder may require explicit channel mapping flags to handle the layout correctly.