Convert CAVS to AIF — Free Online Tool
Extract and convert audio from CAVS video files into AIF format — Apple's lossless, uncompressed audio container. The audio stream is re-encoded from AAC to 16-bit big-endian PCM (pcm_s16be), producing a full-fidelity AIF file suitable for professional audio work on Mac systems.
to
FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
CAVS files encode audio using AAC (Advanced Audio Coding), a lossy compressed format. During this conversion, FFmpeg decodes the AAC audio stream from the CAVS container and re-encodes it as raw 16-bit signed big-endian PCM audio, which is the standard uncompressed format inside an AIF container. The video stream is discarded entirely, since AIF is a pure audio format. Because the source AAC audio is lossy, the output AIF file will be lossless and uncompressed, but its fidelity ceiling is still determined by what was preserved in the original AAC encoding — converting to AIF does not recover audio detail lost during CAVS's original AAC compression. The resulting AIF file will be significantly larger than the source audio because PCM stores every sample as raw data with no compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing tool. All conversion logic — demuxing the CAVS container, decoding AAC audio, encoding PCM, and writing the AIF file — is handled by this single executable, running in your browser via WebAssembly on this tool. |
-i input.cavs
|
Specifies the input file, a CAVS-format video file. FFmpeg reads this file, identifies the AAC audio stream inside the CAVS container, and prepares it for decoding. The video stream is also detected but will be ignored since AIF cannot hold video. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used inside AIF files. This instructs FFmpeg to decode the AAC audio from the CAVS file and re-encode it as raw PCM samples in the byte order that Apple's AIF specification requires. |
output.aif
|
Defines the output file as an AIF container. FFmpeg infers the AIF format from the .aif file extension and wraps the pcm_s16be audio stream in a valid AIF file that Mac applications and professional audio tools can open directly. |
Common Use Cases
- Extracting dialogue or narration from CAVS broadcast content to edit in a Mac-based DAW like Logic Pro or GarageBand, which natively read AIF files
- Archiving the audio track from a Chinese broadcast CAVS recording into an uncompressed format for long-term preservation in a professional media library
- Preparing audio extracted from CAVS video for mastering workflows where uncompressed source files are required by the studio
- Converting CAVS soundtrack audio to AIF so it can be imported into Final Cut Pro or other Apple editing tools that expect AIF for audio assets
- Stripping audio from CAVS content for analysis or forensic purposes where a lossless, uncompressed representation is needed for waveform inspection
- Delivering audio extracted from CAVS-format video to a client on an all-Mac post-production pipeline that standardizes on AIF for audio interchange
Frequently Asked Questions
No. The audio in a CAVS file is encoded with AAC, which is a lossy format that permanently discards some audio data during compression. Converting to AIF re-encodes that decoded AAC audio as uncompressed PCM, which means the AIF file is lossless at the bit level, but it cannot restore the detail that AAC already removed. The AIF output is a lossless copy of whatever fidelity the AAC encoding preserved — not an improvement over the original source.
AIF stores audio as raw, uncompressed PCM data with no compression whatsoever. At 16-bit stereo 44.1 kHz, that amounts to roughly 10 MB per minute of audio, and even more at higher sample rates. The CAVS file's AAC audio, by contrast, is highly compressed — typically at 128 kbps or lower — so the AIF output can easily be 5–10 times larger than the audio portion of the source file. This size increase is expected and is the direct cost of lossless, uncompressed storage.
Yes. FFmpeg decodes the AAC stream from the CAVS file and encodes it into PCM at the same sample rate and channel configuration detected in the source, without resampling or downmixing unless you explicitly add flags to change this. If the CAVS audio is stereo at 48 kHz, the output AIF will also be stereo at 48 kHz. You can verify this in the FFmpeg output log, which reports the detected audio parameters before and after conversion.
Yes. The default command uses pcm_s16be (16-bit), but AIF also supports 24-bit and 32-bit PCM. To get 24-bit output, replace '-c:a pcm_s16be' with '-c:a pcm_s24be' in the FFmpeg command. For 32-bit float output, use '-c:a pcm_f32be'. Higher bit depths will produce larger files but give more headroom for editing without clipping, which is why 24-bit is common in professional audio workflows.
On the command line, you can use a shell loop. On macOS or Linux, run: for f in *.cavs; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.cavs}.aif"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". Each file is processed sequentially, extracting and converting its audio track to a separate AIF file. The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for batch workflows.
CAVS containers carry minimal metadata, and AIF's metadata support is also limited compared to formats like FLAC or MP4. FFmpeg will attempt to map any title, artist, or comment tags it finds in the CAVS file to the AIF output, but in practice CAVS broadcast files rarely contain rich metadata. Do not expect embedded chapter markers or multiple audio track information to survive — AIF does not support either of those features.
Technical Notes
This conversion involves two distinct operations: demuxing the CAVS container to extract the AAC audio stream, and transcoding that AAC audio into 16-bit big-endian PCM for storage in an AIF container. The big-endian byte order of pcm_s16be is a deliberate design choice in the AIF specification, which Apple based on the Motorola/PowerPC byte order convention — this distinguishes AIF from WAV, which uses little-endian PCM. CAVS (GB/T 20090) is primarily a Chinese national broadcast standard and is not widely supported outside of that ecosystem, meaning CAVS files encountered in the wild are often from television or streaming sources encoded with relatively constrained AAC bitrates. Because AIF lacks support for subtitles, chapters, and multiple audio tracks, any secondary audio tracks in a multi-stream CAVS file will be ignored; FFmpeg selects the first audio stream by default. If a specific audio stream is needed, the '-map' flag can be added to the command to target it explicitly. The output AIF file is fully compatible with macOS system tools, Logic Pro, GarageBand, Final Cut Pro, and any other software that reads standard PCM AIF.