MyMediaTools

Convert CAVS to AIFC — Free Online Tool

Extract and convert audio from CAVS video files into AIFC format using PCM 16-bit big-endian encoding — a professional lossless audio format compatible with Apple and broadcast workflows. Since CAVS files carry AAC audio, this tool transcodes that compressed audio stream into uncompressed PCM, giving you a clean, studio-ready AIFC file directly in your browser.

FFmpeg Command

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Estimated output:

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How It Works

CAVS files store video encoded with H.264 and audio encoded as AAC — a lossy compressed format. Converting to AIFC discards the video stream entirely and decodes the AAC audio back to raw PCM samples, which are then written in 16-bit signed big-endian PCM format (pcm_s16be) inside an AIFC container. AIFC is an extension of Apple's AIFF format designed to carry compressed or uncompressed audio; in this case the output is uncompressed PCM, making it lossless from the decoded AAC signal onward. Note that because the source AAC audio was already lossy, the resulting AIFC file captures the best possible quality from what was in the CAVS file, but cannot recover detail lost during the original AAC encoding. The output file will be substantially larger than the source audio due to the uncompressed nature of PCM.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg multimedia processing engine. In this browser-based tool, FFmpeg runs as a WebAssembly binary (FFmpeg.wasm) entirely within your browser, with no server upload required.
-i input.cavs Specifies the input file in CAVS (Chinese Audio Video Standard) format, which contains an H.264 video stream and an AAC audio stream. FFmpeg demuxes both streams from the CAVS container, though only the audio stream is used in this conversion.
-c:a pcm_s16be Sets the audio codec to 16-bit signed PCM in big-endian byte order, which is the standard uncompressed audio encoding for AIFC/AIFF files. This decodes the lossy AAC audio from the CAVS source and re-encodes it as uncompressed PCM — the native, lossless format expected inside an AIFC container.
-b:a 128k Specifies a target audio bitrate of 128 kbps, which matches the typical bitrate of the AAC source audio in the CAVS file. For uncompressed pcm_s16be output, this flag has no effect on actual output quality or file size — PCM bitrate is fixed by sample rate and bit depth — but it is included for compatibility with the tool's quality parameter interface.
output.aifc Defines the output filename with the .aifc extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an AIFC container — Apple's extended Audio Interchange File Format that supports both compressed and uncompressed audio, widely used in professional macOS audio workflows.

Common Use Cases

  • Extracting a clean uncompressed audio track from a CAVS broadcast recording for use in a professional audio editing session in Logic Pro or Pro Tools, which favor AIFC/AIFF files natively
  • Archiving the audio portion of Chinese digital broadcast content (AVS standard) as uncompressed PCM in AIFC for long-term preservation without further lossy compression
  • Delivering audio extracted from CAVS source material to a post-production facility that requires big-endian PCM AIFC files as part of an Apple-ecosystem finishing pipeline
  • Preparing audio from CAVS video files for waveform analysis or forensic audio examination, where uncompressed PCM in AIFC eliminates any codec artifacts that might obscure the signal
  • Converting the audio track of a CAVS-encoded Chinese broadcast clip into AIFC so it can be imported into GarageBand or older Apple audio tools that parse AIFC natively but not AAC-in-CAVS
  • Stripping video from a CAVS file and producing an AIFC master audio file before re-encoding to other formats, using AIFC as an intermediate lossless step in the production chain

Frequently Asked Questions

The AIFC file itself is written as uncompressed 16-bit PCM, which is lossless in the sense that no further quality degradation occurs during this conversion. However, since the audio in the CAVS file was already AAC-encoded (a lossy format), any detail lost at that earlier stage cannot be recovered. The AIFC output represents the highest fidelity you can obtain from the source material, frozen in an uncompressed container.
The AAC audio track inside a CAVS file is heavily compressed — a typical 128 kbps AAC stream is roughly 11x smaller than the equivalent uncompressed PCM. When this tool converts to AIFC with pcm_s16be, it writes raw uncompressed samples at CD-quality resolution (16-bit, big-endian), which expands the audio data dramatically. Additionally, the video stream from the CAVS file is dropped entirely, so the output contains only audio, but that audio is uncompressed and therefore larger per second than the compressed AAC source.
AIFF (Audio Interchange File Format) supports only uncompressed PCM audio, while AIFC (AIFF-Compressed) extends the format to optionally support compressed codecs like MACE or A-law in addition to PCM. In this conversion, the output uses pcm_s16be inside AIFC, which is functionally very similar to a standard AIFF PCM file. Most Apple and professional audio software that reads AIFF will also read AIFC without any distinction, so compatibility is not a concern in practice.
CAVS files are a Chinese broadcast standard container and carry minimal metadata, and AIFC has limited metadata support compared to formats like MP4 or Matroska. This conversion focuses on the audio stream itself; any embedded metadata from the CAVS source may not transfer to the AIFC output. If metadata preservation is critical, you should tag the resulting AIFC file manually using an audio editor after conversion.
You can replace pcm_s16be with pcm_s24be for 24-bit output or pcm_s32be for 32-bit output, which are also valid AIFC codecs and provide greater dynamic range headroom — useful for professional post-production. For example: ffmpeg -i input.cavs -c:a pcm_s24be output.aifc. Keep in mind that since the source audio is 128k AAC, the practical benefit of going above 16-bit is minimal; the extra bit depth will not recover detail that was never in the source.
Yes. On Linux or macOS you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.cavs}.aifc"; done. On Windows PowerShell: Get-ChildItem *.cavs | ForEach-Object { ffmpeg -i $_.FullName -c:a pcm_s16be -b:a 128k ($_.BaseName + '.aifc') }. This is especially useful for processing collections of CAVS broadcast recordings larger than 1GB, which exceed the browser tool's file size limit.

Technical Notes

The pcm_s16be codec used in this conversion writes 16-bit signed integers in big-endian byte order, which is the native byte order for AIFF/AIFC files (originating from Motorola 68k-based Apple hardware). This contrasts with the little-endian PCM used in WAV files on x86 systems. The resulting AIFC file will be fully compatible with macOS Core Audio, Logic Pro, Pro Tools, and any DAW that supports AIFF. The -b:a 128k flag is passed but has no functional effect on uncompressed PCM output — bit rate is determined entirely by sample rate, bit depth, and channel count in lossless PCM, not by a target bitrate parameter. The source AAC audio in CAVS files is typically stereo at 44.1 kHz or 48 kHz; FFmpeg preserves the original sample rate and channel count unless you explicitly specify -ar or -ac flags. CAVS does not support subtitles, chapters, or multiple audio tracks, so there is no risk of losing those elements — the conversion is straightforwardly a single audio stream extraction and PCM re-encoding.

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