Convert WMV to AIFC — Free Online Tool
Convert WMV video files to AIFC audio, extracting and transcoding the WMA (wmav2) or AAC audio track into a big-endian 16-bit PCM AIFC file using the pcm_s16be codec. Ideal for pulling broadcast-quality, uncompressed audio from Windows Media Video sources for use in professional Apple and Pro Tools workflows.
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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) or occasionally AAC, wrapped in Microsoft's Advanced Systems Format (ASF) container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track, then re-encodes it as 16-bit signed big-endian PCM (pcm_s16be) — the default and most universally supported codec for AIFC. The result is a lossless-quality PCM audio file in Apple's AIFC container, which extends the classic AIFF format to support compressed variants. Because WMV audio is lossy (WMA or AAC), the output AIFC file represents a lossless container holding audio that was already lossy at the source — no further generational quality loss occurs beyond the initial decode, but the original compression artifacts from the WMV source are preserved.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.wmv
|
Specifies the input file: a WMV video in Microsoft's Advanced Systems Format (ASF) container, typically carrying a video stream encoded with msmpeg4 and an audio stream encoded with wmav2 or AAC. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio codec for AIFC files and ensures broad compatibility with Apple and professional audio tools such as Pro Tools and Logic Pro. |
-b:a 128k
|
Nominally sets an audio bitrate target of 128 kbps; in practice this flag has no effect on PCM codecs like pcm_s16be because PCM bitrate is fixed by sample rate, bit depth, and channel count rather than a compression target. |
output.aifc
|
Defines the output filename and triggers FFmpeg to use the AIFC muxer, producing an Apple AIFC container with big-endian PCM audio and no video stream — the WMV video track is automatically dropped since AIFC is an audio-only format. |
Common Use Cases
- Importing a Windows Media Video recording into Pro Tools or Logic Pro, which work natively with AIFC/AIFF files but cannot read ASF/WMV containers
- Extracting narration or voice-over audio from a WMV corporate training video to re-edit in an Apple-centric post-production pipeline
- Archiving the audio track from legacy WMV broadcast recordings into a more durable, platform-neutral PCM format for long-term preservation
- Preparing audio extracted from WMV screencasts or webinars for upload to a platform that requires uncompressed or lightly compressed big-endian PCM audio
- Converting WMV conference call recordings into AIFC so they can be imported directly into GarageBand or Final Cut Pro X without transcoding inside the app
- Stripping the WMA audio from a WMV music video to produce a high-fidelity PCM reference file for A/B quality comparison against a lossless source
Frequently Asked Questions
No. WMV files store audio using lossy codecs — most commonly wmav2 (Windows Media Audio 2) or AAC. When FFmpeg decodes that stream and writes it as pcm_s16be PCM in AIFC, the output is a lossless container, but it carries the same audio fidelity as the compressed source. Any artifacts introduced by the original WMA or AAC encoding are already baked in. The advantage is that no additional quality loss occurs during this conversion, and the AIFC file will not degrade further if re-edited or re-exported from a DAW.
WMV files store audio as compressed WMA or AAC data, which can be 10–20 times smaller than uncompressed PCM. The AIFC output uses pcm_s16be, which is raw, uncompressed audio at 16 bits per sample — approximately 10 MB per minute for a standard stereo 44.1 kHz stream. The video stream is also dropped entirely, but the shift from compressed audio to PCM is the dominant reason AIFC files are significantly larger than their WMV counterparts.
FFmpeg will attempt to map compatible metadata tags from the ASF container to the AIFC output, but coverage is partial. WMV files often embed Microsoft-specific ASF metadata (WM/Title, WM/AlbumArtist, etc.) that does not have direct AIFC equivalents. Common fields like title and artist typically transfer, while more exotic ASF tags or any DRM-related metadata will be dropped. It is advisable to verify tags in your DAW or a tool like MediaInfo after conversion.
Yes. AIFC supports several PCM codecs beyond the default pcm_s16be. To get 24-bit big-endian PCM, replace '-c:a pcm_s16be' with '-c:a pcm_s24be'. For 32-bit integer PCM use pcm_s32be, and for 32-bit float use pcm_f32be. Higher bit depths increase file size but preserve more dynamic range headroom — useful if you plan to process the audio extensively in a DAW before a final mixdown. The full command would look like: ffmpeg -i input.wmv -c:a pcm_s24be output.aifc
WMV (ASF) supports multiple audio streams, but AIFC does not — it carries only a single audio track. By default, FFmpeg selects the first (or highest-quality) audio stream for conversion. If you need a specific track from a multi-track WMV, you can target it explicitly by adding '-map 0:a:1' (for the second audio stream) before the output filename in the command, for example: ffmpeg -i input.wmv -map 0:a:1 -c:a pcm_s16be output.aifc
On macOS or Linux, you can loop over all WMV files in a directory with a single shell command: for f in *.wmv; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.wmv}.aifc"; done. On Windows Command Prompt, use: for %f in (*.wmv) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc". This applies the same conversion settings to every file and names each output after its source WMV. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for bulk jobs.
Technical Notes
AIFC (Audio Interchange File Format Compressed) uses a big-endian byte order inherited from Apple's classic Motorola 68000 architecture, which distinguishes it from WAV (little-endian) and makes it the native format for many Apple and Avid audio tools. The default codec used here, pcm_s16be, produces standard CD-quality 16-bit big-endian PCM — suitable for virtually all professional DAWs that accept AIFF or AIFC. The source WMV container uses Microsoft's Advanced Systems Format (ASF), which requires FFmpeg's '-f asf' demuxer internally; this is handled automatically by FFmpeg when reading WMV files. One known limitation is that WMV files protected with Windows Media DRM cannot be decoded by FFmpeg and will produce an error — only DRM-free WMV files can be converted. The '-b:a 128k' flag in the command is technically ignored by PCM codecs (since PCM bitrate is determined entirely by sample rate, bit depth, and channel count), but it is harmless to include. If your WMV source contains a stereo 44.1 kHz stream, the resulting pcm_s16be AIFC file will have a fixed bitrate of approximately 1,411 kbps regardless of the '-b:a' value specified.