Convert WMV to CAF — Free Online Tool
Convert WMV video files to CAF (Core Audio Format), extracting and transcoding the Windows Media Audio stream into Apple's professional audio container using PCM encoding. Ideal for bringing Windows-sourced audio into macOS or iOS audio workflows where CAF's large file support and high-resolution codec compatibility are required.
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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
WMV files store audio using Windows Media Audio codecs (typically WMAv2) inside Microsoft's Advanced Systems Format (ASF) container. Since CAF is an audio-only container with no video support, FFmpeg discards the video stream entirely during this conversion — no video transcoding occurs. The WMAv2 audio is decoded from its lossy compressed state and re-encoded as 16-bit signed little-endian PCM (pcm_s16le), which is an uncompressed linear audio format. This means the output CAF file will be significantly larger than the source WMV, but the audio is stored without any further lossy compression artifacts layered on top of the original. The ASF container's metadata structure is not directly portable to CAF, so track-level tags from the WMV may not be preserved in the output.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles reading the ASF/WMV container, decoding WMAv2 audio, and writing the output CAF file. In this browser tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) without any server involvement. |
-i input.wmv
|
Specifies the input file as a WMV (ASF container) source. FFmpeg inspects the container to identify the embedded video stream (typically MS-MPEG4) and audio stream (typically WMAv2), then selects the audio stream for decoding while ignoring the video since CAF cannot store video data. |
-c:a pcm_s16le
|
Sets the output audio codec to 16-bit signed little-endian PCM, which is an uncompressed linear audio format natively supported by Apple's Core Audio framework and required for broad compatibility with macOS and iOS audio tools. This decodes the lossy WMAv2 source fully before writing uncompressed samples into the CAF container. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. This flag is effectively unused when the codec is pcm_s16le because PCM is an uncompressed format whose actual bitrate is determined by sample rate and bit depth rather than a configurable bitrate target. It is included in the command for consistency but has no impact on the output. |
output.caf
|
Defines the output filename with the .caf extension, which instructs FFmpeg to use the CAF muxer (Core Audio Format container). The resulting file is a valid CAF file containing a single uncompressed PCM audio stream, ready for use in Logic Pro, GarageBand, Core Audio APIs, or any other Apple-ecosystem audio application. |
Common Use Cases
- Extracting the audio narration or voiceover from a Windows Media-based training video to import into Logic Pro or GarageBand on macOS as a raw PCM track for further editing
- Pulling audio from legacy WMV conference recordings or archived Windows Media streams to use in an Apple-native audio production pipeline that expects CAF input
- Converting WMV-encoded audio content for use with Core Audio APIs in macOS or iOS app development, where CAF with pcm_s16le is a natively supported and efficient format
- Preparing audio extracted from WMV screencasts or webinar recordings for analysis or transcription tools on macOS that accept CAF but not ASF-based formats
- Archiving the audio channel of WMV media files in an uncompressed PCM CAF format to avoid further generational quality loss when re-editing on Apple hardware
- Migrating Windows Media Player audio content from a Windows-era media library into an Apple ecosystem workflow where QuickTime and Core Audio tooling is standard
Frequently Asked Questions
There is an inherent quality ceiling because the source WMV audio is encoded with WMAv2, which is a lossy codec. During conversion, FFmpeg fully decodes the WMAv2 stream and re-encodes it as uncompressed 16-bit PCM in the CAF file. This means no additional lossy compression is applied, but any quality already lost in the original WMAv2 encoding cannot be recovered. The output represents the best possible fidelity from the source — it is not lossless in an absolute sense, but it is lossless relative to what the WMV file contained.
WMV stores audio as WMAv2, a compressed lossy format that can achieve bitrates as low as 64–192 kbps. The output CAF uses pcm_s16le, which is uncompressed PCM audio at CD quality — typically around 1.4 Mbps for stereo 44.1 kHz audio. This is a natural consequence of decompressing the audio into a raw format. If file size is a concern and you are working on macOS, you can modify the FFmpeg command to use the AAC or FLAC codec inside CAF instead, both of which are supported by the CAF container.
Yes. CAF with pcm_s16le is a Core Audio native format and is fully supported by Logic Pro, GarageBand, and other Apple DAWs as an importable audio source. QuickTime Player can also open and play pcm_s16le CAF files. Final Cut Pro can import CAF audio for use as a standalone audio clip. This makes the conversion particularly well suited for macOS-based post-production workflows.
The video stream is silently discarded. CAF is a pure audio container and cannot store video data, so FFmpeg automatically drops the video track when writing to a CAF output. No video re-encoding occurs, which means the conversion completes faster and consumes far less CPU than a video transcoding operation would. If you need to retain the video, you should choose a video-capable output format instead.
To use AAC instead of PCM in the CAF file, replace '-c:a pcm_s16le' with '-c:a aac' and the '-b:a 128k' flag will then control the bitrate meaningfully. For lossless output you can use '-c:a flac', which stores FLAC-compressed audio inside the CAF container and produces smaller files than pcm_s16le without any quality loss relative to the decoded WMAv2 source. The '-b:a' flag has no effect when using FLAC and can be omitted in that case.
Yes. On macOS or Linux you can run a shell loop such as: for f in *.wmv; do ffmpeg -i "$f" -c:a pcm_s16le -b:a 128k "${f%.wmv}.caf"; done. On Windows Command Prompt use: for %f in (*.wmv) do ffmpeg -i "%f" -c:a pcm_s16le -b:a 128k "%~nf.caf". Each WMV file will be processed sequentially, extracting and converting its audio track to a separate CAF file. This is especially useful when processing files larger than 1GB, which exceeds this browser tool's limit but works without restriction on the desktop FFmpeg binary.
Technical Notes
The WMV container (technically ASF — Advanced Systems Format) encodes audio with WMAv2 by default, a proprietary Microsoft lossy codec not natively supported by Apple's Core Audio stack. FFmpeg handles WMAv2 decoding via its built-in libavcodec implementation and outputs clean PCM samples to the CAF muxer. The pcm_s16le codec chosen here produces 16-bit signed little-endian samples, which is the same bit depth as standard CD audio and is the default audio format for CAF in Apple's own tooling. One limitation to be aware of is that ASF metadata fields (such as Title, Author, and Copyright stored in WMV headers) use ASF-specific tag structures that FFmpeg may not fully map to CAF metadata chunks, meaning some tags could be lost or truncated. WMV files that include Digital Rights Management (DRM) protection cannot be decoded by FFmpeg and must be decrypted before conversion. Additionally, if the WMV source contains multiple audio tracks — a capability the ASF container supports — FFmpeg will by default select only the first audio track for the CAF output, since CAF does not support multiple audio streams in a single file. If the WMV source was encoded at a sample rate other than 44.1 kHz or 48 kHz, FFmpeg will pass that sample rate through to the CAF output unchanged, which is generally desirable for preserving the original audio fidelity.