Extract Audio from AVI to AIFC — Free Online Tool
Extract audio from AVI video files and save it as AIFC, encoding the output using PCM signed 16-bit big-endian audio — a professional-grade lossless format used in Apple and broadcast workflows. This is ideal when you need uncompressed or minimally processed audio from legacy AVI recordings for use in professional audio applications.
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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 commonly store audio in lossy formats such as MP3 (libmp3lame) or AAC, interleaved with the video stream inside Microsoft's legacy AVI container. During this conversion, FFmpeg strips the video stream entirely using the -vn flag, extracts the raw audio, and re-encodes it into PCM signed 16-bit big-endian format (pcm_s16be) — the default codec for AIFC. Because the source audio in AVI is typically compressed (MP3 or AAC), this process involves decoding that lossy audio to raw PCM and then writing it into the AIFC container. The result is a lossless PCM file, though the original lossy compression artifacts from the AVI source are already baked into the decoded audio. No video data is written to the output file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion. In the browser, this runs via FFmpeg.wasm, a WebAssembly port of the same tool. |
-i input.avi
|
Specifies the input file — in this case an AVI file, which contains interleaved audio and video streams in Microsoft's legacy RIFF-based container format. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore all video streams from the AVI source. This is essential for audio extraction — without it, FFmpeg would attempt to include video in the output, which AIFC cannot contain. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the native default codec for AIFC files. This produces uncompressed audio data stored in big-endian byte order, as expected by Apple and professional broadcast tools that use the AIFC format. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For uncompressed PCM codecs like pcm_s16be, this parameter does not control compression since PCM has no compression — the actual bitrate is fixed by the sample rate and bit depth of the audio stream. |
output.aifc
|
Defines the output filename with the .aifc extension, which tells FFmpeg to write the result into an AIFC container. The .aifc extension distinguishes this from a plain .aiff file and signals that the container may carry compressed or big-endian PCM audio streams. |
Common Use Cases
- Recovering audio from archival or legacy AVI recordings (such as old DV camera exports or early 2000s screen captures) for import into professional Digital Audio Workstations like Logic Pro or Pro Tools, which natively support AIFC
- Extracting dialog or narration audio from AVI-formatted video interviews to use as source material in Final Cut Pro or other Apple-ecosystem editing tools that prefer AIFC/AIFF-family formats
- Pulling audio tracks from AVI training videos or lecture recordings to create standalone audio files for transcription, where an uncompressed PCM format ensures no additional quality degradation during processing
- Converting AVI audio to AIFC as a preparation step before mastering or audio restoration work, since PCM big-endian formats are expected by certain broadcast and professional post-production pipelines
- Archiving the audio component of legacy AVI media collections in a standardized, lossless container format that is more durable and widely supported in professional tools than the original AVI audio stream
Frequently Asked Questions
No — if the audio in the AVI was encoded as MP3 or AAC (which is typical), those compression artifacts are permanently part of the decoded audio signal. Converting to PCM in AIFC captures that decoded audio exactly, without adding new compression, but it cannot recover information that was discarded by the original lossy encoding. The AIFC output will be lossless from this point forward, meaning no further quality degradation occurs in subsequent edits or exports.
The audio in most AVI files is stored in a compressed format like MP3 or AAC, which achieves significant file size reduction. AIFC with pcm_s16be stores uncompressed audio — every sample is written at full resolution with no compression. A 5-minute audio track that might occupy 5–10 MB as MP3 inside an AVI could expand to 50 MB or more as uncompressed 16-bit PCM in AIFC. This size increase is expected and is the tradeoff for lossless, edit-ready audio.
Yes — AIFC supports multiple PCM bit depths including 24-bit (pcm_s24be), 32-bit integer (pcm_s32be), 32-bit float (pcm_f32be), and 64-bit float (pcm_f64be). To use 24-bit for example, you would change the codec flag in the FFmpeg command to '-c:a pcm_s24be'. However, since the AVI source audio is typically 16-bit MP3 or AAC, encoding to higher bit depths will not recover lost detail — it simply provides more headroom for downstream processing in professional workflows that require it.
No — AIFC is a single-stream audio container and does not support multiple audio tracks. AVI can technically carry multiple audio streams, but this is rare in practice. If your AVI has multiple audio tracks, this conversion will extract only the first (default) audio stream. If you need to extract a specific non-default track from an AVI with multiple audio streams, you would modify the FFmpeg command to add '-map 0:a:1' (for the second audio track) before the output filename.
The '-b:a 128k' flag in the command sets a target bitrate, but for PCM codecs like pcm_s16be, bitrate is actually determined by the sample rate and bit depth — not a compression target. PCM is uncompressed, so the -b:a flag has no meaningful effect on the output quality or file size when using pcm_s16be. To influence audio quality meaningfully, you would instead change the sample rate using '-ar 44100' or '-ar 48000', or switch to a higher bit-depth PCM codec like pcm_s24be.
Yes — on the command line you can run this as a shell loop. On Linux or macOS: 'for f in *.avi; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.avi}.aifc"; done'. On Windows PowerShell: 'Get-ChildItem *.avi | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a pcm_s16be -b:a 128k ($_.BaseName + ".aifc") }'. The browser-based tool on this page processes one file at a time, but the FFmpeg command displayed is exactly what runs locally, making it straightforward to adapt for batch use.
Technical Notes
AVI uses Microsoft's RIFF-based interleaved container, where audio and video frames are stored in alternating chunks. The audio codec inside AVI is most commonly MP3 (encoded via libmp3lame) or AAC, though libvorbis is also possible. AIFC (Audio Interchange File Format Compressed) is Apple's extension of AIFF, distinguished by its ability to store both uncompressed PCM and compressed audio streams. The default codec used here — pcm_s16be — is signed 16-bit big-endian PCM, which is the same byte order used in the original AIFF specification and is native to classic Mac OS and many professional broadcast systems. Big-endian byte order differs from the little-endian PCM common in WAV files, which is worth noting if you intend to open the AIFC in software that does not explicitly support big-endian PCM. AIFC does not support chapters, subtitles, or embedded video, and it carries only a single audio stream — making it a clean, purpose-built audio output. Metadata from the AVI source (such as title or artist tags) is generally not preserved during this conversion, as AVI's metadata model (INFO chunks) does not map directly to AIFC's MARK/INST chunk structure.