Convert AVI to AIF — Free Online Tool
Extract and convert audio from an AVI video file into a lossless AIF file using uncompressed PCM audio. This tool discards the video stream and re-encodes whatever audio codec was used in the AVI (commonly MP3 or AAC) into 16-bit big-endian PCM, producing a high-fidelity AIF file ready for use in Apple-native audio workflows.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AVI file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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 as compressed MP3 (libmp3lame) or AAC streams interleaved with video data. During this conversion, FFmpeg reads only the audio stream from the AVI container, discards the video entirely, and decodes the compressed audio to raw PCM samples. Those samples are then re-encoded as 16-bit big-endian signed PCM (pcm_s16be) and wrapped in Apple's AIFF container. Because the source audio in an AVI is almost always lossy (MP3 or AAC), the output AIF file will be lossless and uncompressed — but the audio quality ceiling is determined by the original compressed source, not by AIF's lossless nature. The output file will be significantly larger than the input audio stream because uncompressed PCM at 16-bit stereo requires roughly 10 MB per minute at 44.1 kHz, compared to a few MB for compressed MP3.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles reading the AVI container, decoding its audio stream, and encoding the output AIF file — all running locally in your browser via WebAssembly. |
-i input.avi
|
Specifies the input AVI file. FFmpeg will parse the AVI container to locate its interleaved audio and video streams; only the audio stream will be used in the subsequent steps. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used inside Apple's AIF/AIFF container. This decodes the source AVI audio (typically MP3 or AAC) and re-encodes it as raw PCM samples with no further quality loss. |
output.aif
|
Defines the output filename and signals to FFmpeg that the file should be written as an AIFF container. FFmpeg infers the AIF/AIFF format from the file extension, which is why no explicit -f flag is needed. |
Common Use Cases
- Import legacy AVI video footage into Apple Logic Pro or GarageBand to use its audio track in a music production project, where AIFF is a natively preferred format
- Extract the audio from an old AVI home video or archival recording and store it as a stable, uncompressed AIF file for long-term preservation on macOS
- Pull dialogue or voiceover audio from an AVI file to bring into Final Cut Pro or Pro Tools, both of which handle AIFF natively without requiring additional plugins
- Convert an AVI lecture or presentation recording into AIF so the audio can be edited and cleaned up in a macOS audio editor like Amadeus Pro before republishing
- Deliver raw uncompressed audio extracted from an AVI source to a mastering engineer who requires uncompressed AIFF files as part of their submission workflow
- Strip and convert audio from AVI game recordings or demo files into AIF for use as sound assets in an Apple-platform application or game project
Frequently Asked Questions
No — and this is a common misconception worth understanding. If your AVI file stores audio as MP3 or AAC (which is typical), that audio has already undergone lossy compression. Converting it to AIF wraps the decoded audio in a lossless, uncompressed container, but it cannot recover the detail that was discarded during the original compression. The AIF file will be an exact, uncompressed representation of what remains after the lossy encoding, meaning it preserves current quality with no further degradation but does not restore lost quality.
AVI files typically store audio as compressed MP3 or AAC, which can compress audio by a factor of 10x or more. AIF using pcm_s16be stores every raw audio sample without any compression, so a stereo 44.1 kHz audio track consumes approximately 10 MB per minute. This size increase is expected and is the trade-off for having a lossless, uncompressed file that any audio application can read without a decoder.
The video stream is completely dropped. FFmpeg reads the AVI container, extracts only the audio stream, and writes it into the AIF output. No video data is written to the output file at all, since AIF is a pure audio format with no support for video. If you need to retain the video, you should use a video output format instead.
By default, FFmpeg selects the first audio stream in the AVI file. AIF does not support multiple audio tracks in a single file, so only one stream can be output. If your AVI contains multiple audio tracks (for example, different language dubs), you would need to run separate FFmpeg commands with the -map flag specifying each stream index, such as -map 0:a:1 for the second audio track.
The default command uses pcm_s16be, which produces a 16-bit AIF file — the most compatible option for macOS audio software. If you need higher bit depth, you can replace -c:a pcm_s16be with -c:a pcm_s24be for 24-bit output or -c:a pcm_s32be for 32-bit output. For floating-point audio used in professional DAW workflows, -c:a pcm_f32be is also a valid option. Changing the bit depth will increase the output file size proportionally.
Yes. On macOS or Linux you can use a shell loop: for f in *.avi; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.avi}.aif"; done. On Windows Command Prompt, use: for %f in (*.avi) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". Each AVI file in the directory will have its audio extracted and saved as a separate AIF file with the same base filename. This is especially useful when processing a folder of archival AVI recordings.
Technical Notes
AIF (AIFF) uses big-endian byte ordering for its PCM samples, which is why the codec is named pcm_s16be — signed 16-bit big-endian — rather than the little-endian pcm_s16le used by WAV. This distinction matters when passing the file to low-level audio processing pipelines that read raw bytes, but most audio applications on macOS handle it transparently. AVI's lack of subtitle and chapter support means there is nothing to lose on that front during conversion. One limitation to be aware of is that AVI can encode audio at a wide variety of sample rates, and the AIF output will inherit whatever sample rate was used in the source — no resampling occurs unless explicitly specified with -ar. Metadata embedded in AVI (such as title or artist tags) is generally not preserved in the AIF output because the two containers use incompatible metadata schemes and FFmpeg does not automatically map between them. If audio synchronization artifacts were already present in the AVI (a known issue with some legacy AVI encoders), they will carry through to the AIF since FFmpeg decodes whatever the AVI provides.