Extract Audio from WMV to AIFC — Free Online Tool

Extract audio from WMV video files and save it as AIFC, converting the WMA (wmav2) audio stream into a big-endian PCM format favored by professional audio workflows on Apple platforms. This tool runs entirely in your browser using FFmpeg.wasm — no upload required.

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How It Works

WMV files typically carry audio encoded with Windows Media Audio v2 (wmav2), a lossy proprietary Microsoft codec. During this conversion, FFmpeg discards the video stream entirely and decodes the wmav2 audio to raw PCM samples, then re-encodes them into 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFC container. AIFC is Apple's compressed/uncompressed audio interchange format — a superset of AIFF — and pcm_s16be is its most compatible uncompressed codec, storing audio as signed 16-bit integers in big-endian byte order. Because the output is uncompressed PCM, there is a one-way quality ceiling: any quality lost during the original WMV lossy compression cannot be recovered, but no additional lossy degradation is introduced in this step.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg application — in this browser-based tool, it runs as a WebAssembly (FFmpeg.wasm) binary compiled to execute entirely within your browser tab, with no server involvement.
-i input.wmv Specifies the input file as a WMV (Windows Media Video) using the Advanced Systems Format container. FFmpeg will demux both the msmpeg4/msmpeg4v2 video stream and the wmav2 audio stream from this file.
-vn Disables video output entirely — since this is an audio extraction tool, the msmpeg4 video stream from the WMV is discarded rather than transcoded, keeping the output file as a pure audio AIFC file.
-c:a pcm_s16be Sets the output audio codec to signed 16-bit big-endian PCM, the native uncompressed audio format for AIFC files. This decodes the lossy wmav2 WMA audio from the WMV and re-encodes it as raw PCM samples in big-endian byte order, as expected by the AIFC specification.
-b:a 128k Specifies a 128 kbps audio bitrate target. For PCM codecs like pcm_s16be, the actual bitrate is determined by sample rate and bit depth rather than compression, so this flag has minimal effect on uncompressed PCM output but is included for compatibility with any codec path that may use it.
output.aifc Defines the output filename and tells FFmpeg to wrap the pcm_s16be audio stream in an AIFC container — Apple's Audio Interchange File Format C (Compressed) extension, which is natively readable by Logic Pro, GarageBand, Final Cut Pro, and other Apple audio tools.

Common Use Cases

  • Bringing a Windows Media Video recording (such as a legacy webinar or corporate training video) into a macOS or Logic Pro audio editing session as an AIFC source file
  • Extracting narration or voiceover audio from WMV screencasts to edit in GarageBand or Final Cut Pro, which natively support AIFC/AIFF
  • Archiving the audio track of old WMV home videos in an uncompressed PCM format to prevent further generational quality loss during future editing
  • Pulling a music or soundtrack track out of a WMV promotional video for use in a professional Digital Audio Workstation that requires big-endian PCM interchange files
  • Converting a WMV lecture recording's audio into AIFC so it can be imported into audio analysis or transcription software that does not accept Windows Media formats
  • Stripping the audio from a DRM-free WMV broadcast clip to create a standalone AIFC reference file for audio quality comparison or forensic analysis

Frequently Asked Questions

No. The source audio in a WMV file is encoded with WMA v2 (wmav2), which is a lossy codec — meaning some audio information was permanently discarded when the WMV was originally created. This conversion decodes that lossy audio to PCM and stores it in an uncompressed AIFC container, which faithfully preserves whatever quality remains. You get a lossless container holding a losslessly-decoded version of lossy source material, so quality cannot exceed what the original WMV captured.
AIFC with pcm_s16be stores audio as raw, uncompressed 16-bit samples — no data reduction is applied. WMA v2 audio in a WMV is typically compressed at 64–192 kbps, while 16-bit stereo PCM at 44.1 kHz consumes roughly 1,411 kbps. This 7–20x bitrate difference is completely normal and expected; the larger AIFC file is storing the same audio duration with no compression applied to the output audio stream.
Yes. AIFC with pcm_s16be is one of the most natively supported audio formats across Apple's entire professional software ecosystem. GarageBand, Logic Pro X, Final Cut Pro, and QuickTime all read AIFC files without needing additional codecs or plugins. This is precisely the scenario AIFC was designed for — interchanging audio between Apple-platform professional tools.
Big-endian refers to the byte order used to store each 16-bit audio sample — the most significant byte is stored first. AIFC was originally designed on Motorola 68k and PowerPC hardware, which are big-endian architectures, so big-endian PCM is the native and expected format for AIFC files. Modern Apple software and most professional DAWs handle this byte order transparently, so it does not affect playback compatibility on any current platform.
To use a higher bit-depth PCM codec, replace pcm_s16be with pcm_s24be (24-bit) or pcm_s32be (32-bit) in the -c:a flag — for example: ffmpeg -i input.wmv -vn -c:a pcm_s24be output.aifc. The -b:a flag has limited effect on PCM codecs since their bitrate is determined by sample rate and bit depth rather than compression, but you can also add -ar 48000 to resample to 48 kHz if your professional audio workflow requires it. For lossy AIFC output you could specify pcm_alaw or pcm_mulaw, though those are telecommunications codecs rarely used in modern production.
WMV files store metadata using the Advanced Systems Format (ASF) metadata scheme, while AIFC uses its own chunk-based metadata structure derived from AIFF. FFmpeg will attempt to map common metadata fields such as title, artist, and album between the two containers, but proprietary WMV/ASF tags and any Windows Media DRM metadata will not transfer. Embedded album art is also unlikely to survive this container change, so you may need to re-apply metadata tags using a tool like Kid3 or bwfmetaedit after conversion.

Technical Notes

WMV uses the Advanced Systems Format (ASF) container, which FFmpeg addresses internally with the -f asf flag when writing (not needed here since we are reading). The default WMV audio codec is wmav2, a proprietary Microsoft lossy codec that FFmpeg can decode but not fully emulate in quality terms — decoding is accurate but the original encoder's psychoacoustic model cannot be reconstructed. The output AIFC container supports multiple PCM variants; this tool defaults to pcm_s16be (signed 16-bit big-endian), which is the AIFC standard baseline and provides CD-quality resolution. If the source WMV audio was sampled at a non-standard rate (e.g., 22050 Hz for low-bitrate encodes), the AIFC output will inherit that sample rate unless you explicitly resample with -ar. AIFC does not support multiple audio tracks, so if the WMV contains multiple audio streams, only the first (default) stream will be extracted. No subtitle or chapter data is applicable to either format in this conversion path.

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