MyMediaTools

Extract Audio from CAVS to AIF — Free Online Tool

Extract lossless PCM audio from CAVS video files and save it as AIF — Apple's uncompressed Audio Interchange File Format. This tool strips the AAC-encoded audio track from the Chinese broadcast standard container, decodes it, and re-encodes it as 16-bit big-endian PCM, giving you a pristine, uncompressed audio file ready for use in macOS audio software.

FFmpeg Command

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Estimated output:

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How It Works

CAVS files use AAC as their audio codec, which is a lossy compressed format. During this conversion, FFmpeg discards the video stream entirely (the CAVS/H.264 video is not decoded or re-encoded) and focuses solely on the audio. The AAC audio is decoded from its compressed form back to raw PCM samples, then re-encoded using the pcm_s16be codec — signed 16-bit big-endian PCM — and wrapped in an AIF container. Because AAC is lossy, the original compression artifacts are already baked in; the resulting AIF file is lossless from this point forward, but it cannot recover quality lost during the initial AAC encoding of the CAVS source. The AIF output will be substantially larger than the CAVS source audio because PCM is uncompressed.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg multimedia processing engine. In the browser tool, this runs via FFmpeg.wasm compiled to WebAssembly — the same command works identically on a local desktop FFmpeg installation.
-i input.cavs Specifies the input CAVS file. FFmpeg reads the CAVS container and identifies its streams: an H.264 video track and an AAC audio track. Both are detected but only the audio will be processed.
-vn Disables video output entirely, instructing FFmpeg to ignore the H.264 video stream from the CAVS file. Without this flag, FFmpeg would attempt to include video in the output, which AIF — a pure audio format — cannot contain.
-c:a pcm_s16be Sets the audio codec for the output to signed 16-bit big-endian PCM, the standard uncompressed audio encoding for AIF files on Apple platforms. This decodes the AAC audio from the CAVS source and re-encodes it as raw PCM samples with no further compression.
output.aif Specifies the output filename and container format. The .aif extension tells FFmpeg to wrap the pcm_s16be audio stream in an Audio Interchange File Format container, producing a file natively compatible with macOS audio applications like Logic Pro, GarageBand, and QuickTime.

Common Use Cases

  • Extracting dialogue or narration audio from Chinese broadcast television recordings in CAVS format for use in Final Cut Pro or Logic Pro on a Mac
  • Pulling the audio track from a CAVS-encoded broadcast clip to import into a macOS Digital Audio Workstation (DAW) that requires uncompressed AIF input
  • Archiving the audio portion of CAVS broadcast content in a lossless, widely-compatible container for long-term preservation without further transcoding
  • Isolating sound design elements or music beds from CAVS broadcast masters to use in audio post-production workflows that depend on uncompressed PCM
  • Converting CAVS audio to AIF for delivery to a mastering engineer or studio that requires uncompressed Apple-format audio files
  • Extracting and preserving audio from CAVS files received from Chinese broadcasters or production partners for integration into international post-production pipelines

Frequently Asked Questions

The AIF output is lossless in the sense that once the audio is written as uncompressed PCM, no further quality degradation occurs. However, AAC is a lossy codec, so the audio in your CAVS source has already undergone compression. The conversion process faithfully decodes the AAC stream and preserves exactly what was in the CAVS file — it cannot reconstruct audio detail that AAC discarded during original encoding. If the CAVS source had a high AAC bitrate (e.g., 192k or 320k), the quality loss from AAC is typically minimal and often inaudible.
CAVS files store audio as compressed AAC, which is highly space-efficient. AIF with pcm_s16be stores audio as raw, uncompressed 16-bit samples at the full sample rate. A typical AAC stream at 128k might be 10–15x smaller than the equivalent uncompressed PCM audio. For a 60-minute CAVS broadcast, this could mean an AIF output of several hundred megabytes even though the video (which is discarded) accounted for most of the original file's size.
pcm_s16be stands for signed 16-bit big-endian PCM — the same bit depth and byte order used by standard Audio CD and the default for AIF files on Apple platforms. It is compatible with virtually all macOS audio software including Logic Pro, GarageBand, and Final Cut Pro. If you need higher bit depth (e.g., 24-bit for professional audio work), you can modify the FFmpeg command to use pcm_s24be instead, which AIF also supports. For most CAVS broadcast content mastered at standard quality, 16-bit is sufficient.
FFmpeg will preserve the original sample rate and channel layout (mono, stereo, or surround) from the CAVS AAC stream when writing to AIF. However, CAVS does not support rich metadata, and AIF has limited metadata support compared to formats like FLAC or MP4. Embedded tags such as track title, artist, or album from the CAVS source are unlikely to transfer. If metadata preservation is critical, you may want to add it manually in an audio editor after conversion.
To change the output bit depth, replace pcm_s16be in the command with another PCM variant supported by AIF: pcm_s24be for 24-bit (common in professional audio), pcm_s32be for 32-bit integer, or pcm_f32be for 32-bit float. For example: ffmpeg -i input.cavs -vn -c:a pcm_s24be output.aif. There is no bitrate flag needed for PCM codecs since they are uncompressed — the file size is determined entirely by sample rate, bit depth, and duration.
Yes, you can adapt the command for batch processing in a terminal. On macOS or Linux, use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.cavs}.aif"; done. On Windows Command Prompt: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif". The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for bulk conversions of large CAVS libraries.

Technical Notes

CAVS (Chinese Audio Video Standard) uses AAC as its audio codec and H.264 (libx264) for video. When extracting audio to AIF, the H.264 video stream is completely ignored — FFmpeg never decodes a single video frame, making the process fast regardless of video resolution or duration. The AAC audio is fully decoded before being re-encoded as pcm_s16be; this is a full transcode of the audio stream, not a stream copy, because AAC and PCM are fundamentally different formats. AIF uses big-endian byte order by convention (inherited from Apple's legacy PowerPC architecture), which is why pcm_s16be (big-endian) is the default rather than the little-endian pcm_s16le used in WAV. Most modern macOS applications handle AIF natively without any compatibility issues. One known limitation: CAVS does not support multiple audio tracks, so there is always exactly one audio stream to extract. If the CAVS source was encoded at a low AAC bitrate (e.g., 64k), the resulting AIF will accurately represent that compressed audio with all its artifacts preserved — the uncompressed container cannot compensate for lossy source encoding.

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