Convert AVI to AIFF — Free Online Tool
Extract and convert the audio track from an AVI video file into a high-quality, uncompressed AIFF file using PCM 16-bit big-endian encoding. AIFF is Apple's professional lossless audio container, making this conversion ideal for bringing legacy AVI audio into macOS workflows, DAWs, or any application demanding uncompressed fidelity.
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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
AVI files typically store audio using lossy codecs such as MP3 (libmp3lame) or AAC — the audio is already encoded and compressed inside the container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track from the AVI file back to raw PCM samples, then re-encodes those samples as 16-bit big-endian PCM (pcm_s16be) and wraps them in an AIFF container. Because AIFF is uncompressed and lossless, no further quality is lost beyond whatever was already lost when the original AVI audio was encoded. The result is a full-fidelity, uncompressed audio file ready for professional use.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the AVI container and audio stream, and encoding the output AIFF file. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) with no server upload. |
-i input.avi
|
Specifies the input file, which is an AVI container. FFmpeg will demux this file, identifying the video and audio streams inside it; for this conversion, only the audio stream will be used. |
-c:a pcm_s16be
|
Sets the audio codec for the output to PCM signed 16-bit big-endian, the standard uncompressed encoding used by AIFF. This decodes whatever lossy audio codec was in the AVI (commonly MP3 or AAC) and re-encodes it as raw, uncompressed PCM samples suitable for Apple's AIFF container. |
output.aiff
|
Defines the output filename and, by its .aiff extension, instructs FFmpeg to write the result as an Audio Interchange File Format container. FFmpeg uses the file extension to select the correct muxer, so the .aiff extension is essential for producing a valid AIFF file. |
Common Use Cases
- Extracting the audio commentary or narration from an archival AVI video to import into a macOS Logic Pro or GarageBand session as an uncompressed AIFF track
- Pulling the music or score from an old AVI film clip to use as a lossless audio asset in a Final Cut Pro project, which natively favors AIFF
- Converting AVI-sourced audio to AIFF for mastering workflows where downstream tools require uncompressed PCM audio in a big-endian format
- Archiving the audio track of legacy AVI home videos in an uncompressed format so it can be losslessly re-encoded to any target format in the future without generational quality loss
- Extracting game cutscene audio stored in AVI files for use as sound effects or music cues in a macOS audio production environment
- Preparing audio from AVI broadcast recordings for submission to a client or broadcaster that mandates uncompressed AIFF deliverables
Frequently Asked Questions
No — converting to AIFF will not restore quality that was lost when the original AVI audio was encoded with a lossy codec like MP3 or AAC. What AIFF guarantees is that no additional quality is lost during or after this conversion, since it stores raw uncompressed PCM samples. The output will be a perfect, uncompressed representation of whatever fidelity existed in the AVI's audio track.
AIFF stores audio as uncompressed PCM data, which is inherently much larger than the lossy-compressed audio (MP3 or AAC) typically found in AVI files. A 128 kbps MP3 stream might consume around 1 MB per minute, whereas 16-bit stereo PCM at 44.1 kHz uses roughly 10 MB per minute. The size increase is expected and is the cost of storing audio in an uncompressed, lossless format.
The video stream is completely dropped. AIFF is a pure audio container with no support for video data, so FFmpeg extracts only the audio track from the AVI file. The resulting AIFF file contains no video whatsoever. If you need to keep the video, you should use a different output format such as MP4 or MKV.
Yes. Replace pcm_s16be with pcm_s24be in the command to get 24-bit big-endian PCM, which offers a higher dynamic range and is preferred in professional audio mastering contexts. You can also use pcm_s32be for 32-bit integer or pcm_f32be for 32-bit float PCM. The command would then read: ffmpeg -i input.avi -c:a pcm_s24be output.aiff
By default, FFmpeg selects the first audio track in the AVI file. Since AIFF does not support multiple audio tracks, only one stream can be written to the output. If you need to extract a specific audio track other than the first, add the flag -map 0:a:1 (for the second audio track) to the command before the output filename, adjusting the index as needed.
On Linux or macOS, you can run a shell loop: for f in *.avi; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.avi}.aiff"; done. On Windows Command Prompt, use: for %f in (*.avi) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This applies the same conversion command to every AVI file in the current directory and names each output file after its source.
Technical Notes
The default codec chosen for this conversion is pcm_s16be — signed 16-bit PCM with big-endian byte order — which is the canonical audio encoding for AIFF and is natively expected by Apple platforms and most professional audio software on macOS. AVI does not enforce a single audio codec; files in the wild may carry MP3, AAC, AC-3, PCM, or even ADPCM audio, all of which FFmpeg will decode before re-encoding to PCM. Because AVI does not support chapters, embedded subtitles, or rich metadata beyond basic ID3-style tags, very little metadata carries over to the AIFF output — artist, title, and similar fields are generally not preserved. AVI also technically supports multiple audio tracks, but AIFF can hold only one; FFmpeg will default to the first audio stream. One important limitation to be aware of: if the source AVI contains audio encoded at a sample rate other than a standard rate (e.g., 44100 Hz or 48000 Hz), FFmpeg will preserve that sample rate in the AIFF output without resampling, which may cause compatibility issues in some DAWs. Add -ar 44100 to the command to force a standard sample rate if needed.