Extract Audio from AVI to AIFF — Free Online Tool
Extract audio from AVI files and save it as lossless AIFF, converting whatever lossy codec (MP3, AAC, or Vorbis) was embedded in the AVI to uncompressed PCM audio stored in big-endian 16-bit format. Ideal for recovering audio from legacy AVI recordings at the highest possible fidelity without re-encoding to another lossy format.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
AVI files typically carry audio encoded in a lossy format such as MP3 (libmp3lame), AAC, or Vorbis. During this conversion, FFmpeg discards the video stream entirely using the -vn flag, then decodes the compressed audio from the AVI container and re-encodes it into uncompressed PCM signed 16-bit big-endian audio (pcm_s16be), which is the native codec expected by the AIFF container. Because the source audio is lossy, this process performs a full decode-and-re-encode rather than a simple stream copy — the compressed audio is fully decoded to raw PCM first, then written out to AIFF. No additional quality is lost beyond what was already lost when the AVI's audio was originally compressed, and from this point forward the audio is stored without any further compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles all the decoding, stream selection, and encoding work for this AVI-to-AIFF audio extraction. |
-i input.avi
|
Specifies the input AVI file. FFmpeg reads the RIFF-based AVI container and identifies the available audio and video streams inside it, which may include MP3, AAC, or Vorbis audio alongside a video stream. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream from the AVI and produce an audio-only output. This is essential since AIFF is a pure audio format and cannot carry video data. |
-c:a pcm_s16be
|
Encodes the audio using the PCM signed 16-bit big-endian codec, which is the standard uncompressed audio codec for the AIFF format. This decodes whatever lossy audio (MP3, AAC, or Vorbis) was stored in the AVI into raw, uncompressed samples that AIFF stores without further compression. |
output.aiff
|
Sets the output filename and tells FFmpeg to use the AIFF container format, which is inferred from the .aiff file extension. The resulting file is an uncompressed lossless audio file ready for use in macOS audio applications or professional DAWs. |
Common Use Cases
- Importing audio from archival AVI footage into Apple Logic Pro or GarageBand, which natively work with AIFF files on macOS
- Extracting voiceover or dialogue audio from a legacy AVI video project to edit in an audio workstation without adding another generation of lossy compression
- Recovering high-fidelity audio from old AVI screen recordings or tutorials where the video is no longer needed but the spoken content is valuable
- Preparing audio from AVI source files for mastering or broadcast, where downstream tools require uncompressed AIFF rather than compressed MP3 or AAC
- Stripping the audio track from AVI files captured by older DV camcorders or capture cards to archive just the audio in a professional lossless format
- Converting AVI audio to AIFF as a lossless intermediate format before further processing such as sample rate conversion or bit-depth upscaling
Frequently Asked Questions
No — and this is an important distinction. AVI files almost always contain audio that was already encoded with a lossy codec like MP3, AAC, or Vorbis. The conversion decodes that compressed audio to raw PCM and stores it in AIFF's uncompressed format, which means no further quality is lost from this point forward. However, any quality that was lost when the audio was originally compressed inside the AVI cannot be recovered. The AIFF output will be a perfect lossless snapshot of whatever quality the AVI's audio contained.
This is expected. The AVI likely stored its audio as compressed MP3 or AAC at around 128 kbps, whereas the AIFF output stores audio as uncompressed 16-bit PCM at full sample rate — typically around 1,411 kbps for 44.1kHz stereo audio. The AIFF format trades file size for lossless storage, which is why audio professionals use it as an archival or interchange format rather than a delivery format.
Yes, by modifying the FFmpeg command directly. Replace pcm_s16be with pcm_s24be for 24-bit output or pcm_s32be for 32-bit integer output. However, because the source audio in the AVI was likely compressed at 16-bit or lower fidelity to begin with, choosing a higher bit depth will not recover any additional audio information — it simply stores the same decoded audio in a larger container. 24-bit AIFF is worth using if the original AVI audio was recorded from a 24-bit source.
By default, this FFmpeg command extracts only the first audio track from the AVI. AVI does support multiple audio tracks, though it is relatively rare in practice. If your AVI has multiple tracks and you need a specific one, you can add -map 0:a:1 (for the second audio track) or -map 0:a:2 (for the third) before the output filename in the command. AIFF does not support multiple audio tracks in a single file, so each track would need to be extracted into a separate AIFF file.
On macOS or Linux, you can run a shell loop in Terminal: for f in *.avi; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.avi}.aiff"; done. On Windows Command Prompt, use: for %f in (*.avi) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". This will process every AVI file in the current directory and produce a corresponding AIFF file with the same base filename.
AIFF was developed by Apple and is natively supported on macOS and iOS, but Windows does not support it natively through most media players. However, professional audio applications on Windows — including Adobe Audition, Reaper, and Pro Tools — handle AIFF without any issues. If you need a lossless audio file that works broadly on Windows without additional software, WAV (using pcm_s16le, the little-endian equivalent) is a more universally compatible alternative.
Technical Notes
The AVI container uses a legacy RIFF-based structure and its audio streams are most commonly encoded with libmp3lame (MP3), though AAC and Vorbis are also possible depending on the software that created the file. During this conversion, FFmpeg fully decodes the compressed audio to raw PCM before writing it as pcm_s16be into the AIFF container — this is a transcode, not a remux. The pcm_s16be codec writes signed 16-bit samples in big-endian byte order, which is the AIFF standard established by Apple on the original Motorola 68k architecture. One key limitation: AVI does not store extensive metadata (no chapters, no embedded lyrics, minimal tag support), so the AIFF output will likely contain no title, artist, or album metadata beyond what FFmpeg can infer. If the AVI's audio sample rate is not 44100 Hz or 48000 Hz, FFmpeg will preserve the original sample rate in the AIFF output without resampling unless you explicitly specify -ar. AIFF does not support multiple audio tracks per file, so if the source AVI contained multiple audio streams, only the first will be extracted by default.