Convert AIFF to AC3 — Free Online Tool
Convert AIFF audio files to AC3 (Dolby Digital) format using FFmpeg in your browser. This tool transcodes Apple's uncompressed PCM audio into the AC3 codec widely used in DVD, Blu-ray, and broadcast workflows, giving you Dolby Digital-compatible files without needing to install any software.
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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
AIFF stores audio as raw, uncompressed PCM data (typically 16-bit, 24-bit, or 32-bit big-endian samples), so every sample value is preserved exactly as recorded. During this conversion, FFmpeg reads those uncompressed PCM samples and encodes them using the AC3 (Dolby Digital) codec, which applies perceptual audio compression to significantly reduce file size. AC3 is a lossy format — it analyzes the audio and discards frequency content deemed less perceptually significant by the human ear. The output is a raw .ac3 bitstream at 192 kbps by default, which is the standard stereo bitrate for AC3 used in many DVD and broadcast applications. Because AIFF is lossless and AC3 is lossy, this is a one-way quality tradeoff: you cannot recover the original uncompressed PCM data from the AC3 output.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application, the open-source multimedia processing engine that handles the decoding of AIFF's uncompressed PCM audio and the encoding into AC3 format. |
-i input.aiff
|
Specifies the input file — an AIFF file containing uncompressed PCM audio (big-endian, typically 16-bit or 24-bit). FFmpeg automatically detects the specific PCM variant used inside the AIFF container. |
-c:a ac3
|
Tells FFmpeg to encode the audio stream using the AC3 (Dolby Digital) codec, replacing the uncompressed PCM data from the AIFF source with perceptually compressed Dolby Digital audio suitable for DVD, Blu-ray, and broadcast use. |
-b:a 192k
|
Sets the AC3 audio bitrate to 192 kilobits per second, the standard stereo bitrate for Dolby Digital in DVD applications. Increasing this value (e.g., to 384k or 640k) improves quality at the cost of a larger file; 640k is the maximum and is used for 5.1 surround on Blu-ray. |
output.ac3
|
Defines the output filename with the .ac3 extension, which produces a raw Dolby Digital elementary stream. This format is directly importable into DVD and Blu-ray authoring tools, and playable by Dolby Digital-compatible hardware decoders and AV receivers. |
Common Use Cases
- Preparing professionally recorded AIFF audio masters for inclusion in a DVD or Blu-ray authoring project that requires Dolby Digital AC3 streams
- Converting studio-recorded AIFF files to AC3 for broadcast television delivery, where AC3 is a mandated or preferred audio format
- Reducing the storage footprint of large uncompressed AIFF archives for distribution while retaining Dolby Digital compatibility for set-top boxes and AV receivers
- Creating AC3 audio tracks to pair with video in MPEG-2 transport streams or DVD-Video structures where AC3 is the standard companion audio format
- Transcoding AIFF audio produced on macOS into an AC3 file that can be tested for compatibility with consumer Dolby Digital decoders and home theater systems
- Generating AC3 audio for use in video editing timelines that require Dolby Digital tracks, such as Premiere Pro or DaVinci Resolve sequences targeting disc output
Frequently Asked Questions
Yes — AC3 is a lossy codec, so converting from lossless AIFF will introduce some irreversible quality loss. At the default 192 kbps bitrate, the quality is generally considered transparent for most stereo listening, which is why it is the standard rate used on many DVDs. However, because AIFF preserves every PCM sample with full fidelity, audiophiles doing critical listening comparisons may detect subtle differences, particularly in transient-heavy material. If audio quality is paramount, consider using a higher bitrate like 384k or 640k.
AIFF stores audio as completely uncompressed PCM data, so a single minute of 16-bit stereo audio at 44.1 kHz occupies roughly 10 MB. AC3 at 192 kbps compresses that same audio down to about 1.4 MB per minute — an approximately 7:1 reduction. This dramatic size difference comes from AC3's perceptual coding, which exploits psychoacoustic masking effects to discard audio information the human ear is statistically unlikely to notice. The tradeoff is that the compression is lossy and cannot be undone.
No. AIFF can store audio at 16-bit, 24-bit, or even 32-bit float precision, capturing the full dynamic range of professional recording sessions. The AC3 codec operates internally at a fixed precision and does not carry forward raw bit-depth information the way a lossless format would. Regardless of whether your source AIFF is 16-bit or 32-bit, the encoded AC3 output represents audio through its lossy perceptual model, so the concept of bit depth as a quality metric no longer applies in the same way.
In most cases, yes. The output is a raw AC3 bitstream (.ac3) which is the correct elementary stream format expected by DVD authoring applications like DVD Architect, Encore, or similar tools. However, Blu-ray typically requires AC3 at 640 kbps for 5.1 surround or specific channel configurations — if your source AIFF is stereo and you are authoring a Blu-ray, confirm your authoring software's channel and bitrate requirements before converting.
Replace the value after '-b:a' with your desired bitrate. AC3 supports bitrates of 96k, 128k, 192k, 256k, 320k, 384k, 448k, and 640k. For example, to get the highest quality AC3 output, use: ffmpeg -i input.aiff -c:a ac3 -b:a 640k output.ac3. For 5.1 surround sound content, 640k is the standard maximum and is commonly required for Blu-ray compliance. For stereo content in broadcast contexts, 192k or 256k is typically sufficient.
The single-file command shown on this page converts one file at a time, but you can batch process using a shell loop. On Linux or macOS, run: for f in *.aiff; do ffmpeg -i "$f" -c:a ac3 -b:a 192k "${f%.aiff}.ac3"; done. On Windows PowerShell, use: Get-ChildItem *.aiff | ForEach-Object { ffmpeg -i $_.FullName -c:a ac3 -b:a 192k ($_.BaseName + '.ac3') }. This is particularly useful for converting an entire library of AIFF masters to AC3 for a disc authoring project.
Technical Notes
AIFF's audio codec is always a variant of big-endian PCM (pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, or pcm_f64be), and FFmpeg handles all of these transparently as the input source — you do not need to specify the input codec. The AC3 encoder in FFmpeg (the native ac3 encoder) supports stereo and multichannel audio up to 5.1 channels; if your AIFF source is stereo, the output will be stereo AC3. AC3 supports a fixed set of sample rates (32 kHz, 44.1 kHz, and 48 kHz); if your AIFF was recorded at an unusual sample rate, FFmpeg will automatically resample it to the nearest supported AC3 rate, which could introduce minor additional quality changes. Metadata embedded in AIFF files (such as artist, album, or track name tags) is not carried over to raw .ac3 output files, since the AC3 elementary stream format does not have a standard metadata container. If you need to preserve metadata alongside AC3 audio, consider muxing the AC3 stream into an MP4 or MKV container instead of outputting a raw .ac3 file. Additionally, AIFF files with 32-bit float or 64-bit float PCM will be converted through FFmpeg's internal pipeline without issue, as the AC3 encoder accepts float input.