Convert WMV to AIFF — Free Online Tool
Convert WMV video files to AIFF audio by extracting and decoding the WMA-encoded audio stream into uncompressed 16-bit big-endian PCM — Apple's native lossless audio format. This is the ideal path when you need broadcast-ready, uncompressed audio from Windows Media content for use in macOS DAWs or professional audio workflows.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
WMV files typically carry audio encoded with the WMA v2 (wmav2) codec inside an Advanced Systems Format (ASF) container — a lossy, compressed audio format developed by Microsoft. During this conversion, FFmpeg discards the video stream entirely and decodes the wmav2 audio to raw PCM samples, then re-encodes those samples as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFF container. Because AIFF is uncompressed, the output file will be substantially larger than the source WMV. The audio quality ceiling is determined by the original WMV's lossy encoding — AIFF preserves every sample decoded from the WMA stream with perfect fidelity, but it cannot restore detail that was discarded when the WMV was originally encoded.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the WMV/ASF container and wmav2 audio, and encoding the output as AIFF PCM. |
-i input.wmv
|
Specifies the input file as a WMV (Windows Media Video) file. FFmpeg reads the ASF container, identifies the wmav2 audio stream and the msmpeg4 video stream, and makes both available for processing. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — signed 16-bit big-endian PCM — which is the standard uncompressed audio encoding required by the AIFF format specification. This decodes the lossy wmav2 audio from the WMV and writes it as raw, uncompressed samples. |
output.aiff
|
Specifies the output filename with the .aiff extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an Audio Interchange File Format container — Apple's uncompressed lossless audio format natively supported by macOS and professional audio software. |
Common Use Cases
- Importing dialogue or narration from a Windows Media training video into Logic Pro, GarageBand, or Pro Tools for audio cleanup and re-export
- Extracting a music soundtrack from a WMV promotional video to use as an uncompressed source file in a macOS video editing project in Final Cut Pro
- Archiving the audio track from legacy WMV corporate recordings into a format-agnostic, lossless container before the original files become unplayable
- Stripping audio from a Windows Media Player-distributed video to submit to a broadcast facility that requires uncompressed AIFF deliverables
- Preparing audio extracted from a WMV screen recording for forensic analysis or transcription, where exact sample-level accuracy is required
- Converting WMV audio content for use on macOS systems where WMA playback support is absent but AIFF is natively supported by Core Audio
Frequently Asked Questions
No — the audio in a WMV file is encoded with the lossy WMA v2 codec, which permanently discards audio information during the original encoding process. Converting to AIFF decodes the WMA stream into uncompressed PCM and preserves every sample that survived WMA compression, but it cannot reconstruct the detail that was already lost. The result is a lossless container holding a losslessly-decoded version of a lossy source — a common and perfectly valid archival practice, but not a quality upgrade.
WMV stores audio as compressed WMA v2, which typically runs at 128 kbps — a heavily compressed representation. AIFF stores raw, uncompressed PCM samples: 16-bit stereo audio at 44.1 kHz consumes roughly 10 MB per minute, compared to about 1 MB per minute for 128 kbps WMA. The video stream is dropped entirely, but the switch from lossy compression to uncompressed PCM is the primary driver of the size increase for the audio portion.
WMV files can carry ASF-format metadata (title, author, copyright, etc.), but AIFF uses a different metadata structure based on IFF chunks (NAME, AUTH, ANNO). FFmpeg does not reliably map ASF metadata fields to AIFF chunks during this conversion, so metadata is likely to be dropped. If preserving metadata matters, you should note the tags from the source file before converting and re-apply them to the AIFF using a tool like Kid3 or exiftool after conversion.
Yes. The default command uses pcm_s16be (16-bit signed big-endian PCM), which is standard CD quality. You can substitute a higher bit-depth codec by changing the -c:a flag: use pcm_s24be for 24-bit, pcm_s32be for 32-bit integer, or pcm_f32be for 32-bit float. For example: ffmpeg -i input.wmv -c:a pcm_s24be output.aiff. Since the WMV source is lossy WMA, upsampling to 24-bit will not recover lost audio detail, but 24-bit output may be required by certain professional DAW or broadcast workflows.
FFmpeg processes one input file per invocation, so batch conversion requires a shell loop. On macOS or Linux, you can run: for f in *.wmv; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.wmv}.aiff"; done. On Windows Command Prompt, use: for %f in (*.wmv) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This applies the same pcm_s16be conversion to every WMV file in the current directory.
The video stream is implicitly dropped because AIFF is a pure audio format with no container support for video. FFmpeg detects that the output container (AIFF) cannot carry video and automatically excludes all video streams. You do not need to add a -vn flag, though including it explicitly makes the intent clear. Only the first audio track from the WMV is written to the AIFF output, as AIFF does not support multiple audio tracks.
Technical Notes
The WMV container (ASF) in this pipeline carries wmav2 audio, a proprietary Microsoft codec with variable bitrate support, typically encoded between 64 kbps and 192 kbps for standard content. FFmpeg's wmav2 decoder is mature and stable, so decoding artifacts are rarely introduced during conversion. The output codec pcm_s16be specifies signed, 16-bit, big-endian PCM — the byte order required by the AIFF specification (as opposed to the little-endian PCM used in WAV). The sample rate of the output will match whatever rate is present in the source WMV (commonly 44100 Hz or 48000 Hz); FFmpeg does not resample unless explicitly instructed with the -ar flag. AIFF does not support subtitles, chapters, or multiple audio tracks, and the -f asf special flag used when creating WMV files is irrelevant here since we are only reading the WMV. One known limitation: if the WMV file uses DRM (Digital Rights Management) — a supported feature of the ASF/WMV format — FFmpeg will be unable to decode the audio stream and the conversion will fail. DRM-protected WMV files must be decrypted through licensed playback software before any format conversion is possible.