Convert AC3 to ALAC — Free Online Tool
Convert AC3 (Dolby Digital) audio files to ALAC (Apple Lossless Audio Codec) stored in an M4A container. This tool decodes the lossy AC3 stream and re-encodes it into a lossless ALAC format, making your Dolby Digital audio fully compatible with iTunes, Apple Music, and the broader Apple ecosystem without further generational quality loss.
to
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 Dolby Digital codec — the audio was already compressed and some original data was discarded when the AC3 file was created. During this conversion, FFmpeg fully decodes the AC3 bitstream back to raw PCM audio in memory, then re-encodes it using Apple's ALAC codec and wraps the result in an MPEG-4 (.m4a) container. Because ALAC is lossless, the output file is a bit-perfect capture of whatever audio survived the original AC3 encoding — no additional quality is lost in this step. The resulting M4A file will be larger than the AC3 source, since ALAC uses lossless compression rather than the aggressive perceptual coding AC3 relies on. Surround sound channel mappings (such as 5.1) present in the AC3 source are preserved through the decode-encode pipeline.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles decoding the AC3 Dolby Digital input and re-encoding it as ALAC. |
-i input.ac3
|
Specifies the input file — an AC3 Dolby Digital audio file. FFmpeg reads the bitstream and decodes it to raw PCM audio in memory for re-encoding. |
-c:a alac
|
Sets the audio codec for the output to ALAC (Apple Lossless Audio Codec). FFmpeg takes the decoded PCM audio from the AC3 source and compresses it losslessly using Apple's ALAC encoder. |
-c:a alac
|
A duplicate of the preceding flag that has no additional effect — the codec is already set to ALAC. This redundant entry can be safely removed when running the command locally; the conversion behavior is identical without it. |
output.m4a
|
Specifies the output filename with an .m4a extension. The .m4a extension signals FFmpeg to wrap the ALAC audio stream in an MPEG-4 container, which is the standard packaging used by iTunes and Apple devices for ALAC audio files. |
Common Use Cases
- Ripping Dolby Digital audio tracks from a DVD or Blu-ray and archiving them in a lossless format for long-term storage without further quality degradation.
- Importing a 5.1 surround sound AC3 track into an Apple-native editing workflow in Final Cut Pro or Logic Pro, which prefer ALAC or other Apple-compatible formats over raw AC3.
- Adding Dolby Digital broadcast audio to an iTunes or Apple Music library, where AC3 is not natively supported but ALAC M4A files import seamlessly.
- Preserving a film's original AC3 dialogue or effects track in a lossless container before doing audio restoration or post-processing work in an audio editor.
- Converting AC3 audio extracted from a DVR recording or over-the-air broadcast capture into a format playable on iPhone, iPad, or Apple TV without transcoding on the device.
- Creating a lossless master archive of a Dolby Digital audio asset so future format conversions can be made from ALAC rather than from the lossy AC3 source.
Frequently Asked Questions
No — and this is an important distinction. AC3 is a lossy format, meaning audio data was permanently discarded when the AC3 file was originally encoded. Converting it to ALAC does not recover that lost information. What ALAC guarantees is that the audio you hear in the AC3 file is preserved exactly, with no further quality loss introduced by this conversion step. Think of it as freezing the current quality in a lossless container rather than restoring it to its original state.
AC3 achieves its small file size through aggressive perceptual audio compression that permanently discards data the human ear is less likely to notice. ALAC, by contrast, uses lossless compression — it retains every audio sample without any data loss. Lossless compression is far less space-efficient than lossy compression, so the resulting M4A file will typically be several times larger than the AC3 source, especially for multi-channel 5.1 surround audio tracks.
Yes. ALAC supports multi-channel audio including 5.1 surround sound configurations. FFmpeg decodes all AC3 channels — front left, front right, center, LFE (subwoofer), surround left, and surround right — and re-encodes them into the ALAC stream. The channel layout is preserved in the output M4A file. Apple devices and software including iTunes and Final Cut Pro can read multi-channel ALAC audio correctly.
Yes. ALAC inside an M4A container is a first-party Apple format, so it plays natively on iPhone, iPad, Mac, Apple TV, and HomePod without any additional software. iTunes and Apple Music will import the file and display it alongside other library tracks. This is one of the primary reasons to choose this conversion — moving from an AC3 track that Apple software may not support directly into a format with full native Apple compatibility.
The command shown includes '-c:a alac' twice, which is redundant — the second instance simply overwrites the first with the same value, so there is no functional difference. The effective instruction to FFmpeg is a single '-c:a alac', which tells it to encode the audio output using Apple's ALAC codec. If you run this command locally, you can safely remove the duplicate flag: 'ffmpeg -i input.ac3 -c:a alac output.m4a' produces identical results.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.ac3; do ffmpeg -i "$f" -c:a alac "${f%.ac3}.m4a"; done'. On Windows Command Prompt, use: 'for %f in (*.ac3) do ffmpeg -i "%f" -c:a "%~nf.m4a"'. Each AC3 file in the directory will be decoded and re-encoded to a corresponding ALAC M4A file. This is especially useful when archiving an entire collection of Dolby Digital audio tracks extracted from DVD or broadcast recordings.
Technical Notes
AC3 supports bitrates ranging from 32 kbps up to 640 kbps, with 192 kbps and 384 kbps being most common for stereo and 5.1 content respectively. The decoded PCM audio quality is bounded by whichever AC3 bitrate was used during the original encoding — a 192 kbps AC3 source decoded to PCM and then stored in ALAC will sound identical to the AC3, not better. ALAC itself does not have a quality parameter; it is always lossless relative to whatever PCM it encodes. The M4A container produced supports chapter markers (though AC3 source files carry no chapter data to transfer), but does not support embedded subtitle tracks or multiple audio streams. Metadata tags such as title and track number can be added to the output M4A using additional FFmpeg '-metadata' flags if needed. One known limitation: if the AC3 source uses Dolby Digital Plus (E-AC3, a related but distinct codec), it will not be recognized as a plain AC3 file and this command will need to be adjusted accordingly. Always verify channel count and sample rate in the FFmpeg console output to confirm the surround configuration was correctly interpreted during decode.