MyMediaTools

Convert WMA to AIFC — Free Online Tool

Convert WMA (Windows Media Audio) files to AIFC format, decoding the lossy wmav2-encoded audio and re-encoding it into lossless PCM big-endian audio (pcm_s16be) suitable for professional audio workflows on Apple and Unix systems. This is ideal when you need to bring legacy Windows Media audio into a format compatible with professional DAWs and audio editors that prefer the AIFF/AIFC container.

FFmpeg Command

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Estimated output:

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How It Works

WMA files use Microsoft's proprietary wmav2 (or wmav1) lossy codec, which compresses audio using perceptual encoding similar to MP3. During this conversion, FFmpeg fully decodes the WMA audio stream back to raw PCM samples, then re-encodes it into pcm_s16be — 16-bit signed big-endian PCM — wrapped in the AIFC container. Because WMA is lossy, the decoded audio already carries whatever quality loss occurred during the original WMA encoding; the resulting AIFC file preserves that decoded audio faithfully in lossless PCM form, meaning no additional quality is lost in the transcoding step itself. The AIFC container extends Apple's AIFF format to support compressed audio codecs, but in this case the output is uncompressed PCM, making it universally readable by professional audio tools while being significantly larger than the source WMA file.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg multimedia processing tool. In the browser-based version of this tool, this runs via FFmpeg.wasm compiled to WebAssembly, executing entirely in your browser without any server upload.
-i input.wma Specifies the input file — a WMA audio file encoded with Microsoft's wmav2 (or wmav1) lossy codec inside an ASF container. FFmpeg reads and decodes this proprietary audio stream before any output processing begins.
-c:a pcm_s16be Sets the output audio codec to 16-bit signed big-endian PCM, which is the default and most compatible uncompressed format for the AIFC container. This replaces the lossy wmav2 compression with lossless raw PCM samples, making the audio suitable for professional audio editing tools that expect uncompressed data.
-b:a 128k Specifies a target audio bitrate of 128 kbps. For uncompressed PCM codecs like pcm_s16be, the actual bitrate is determined by sample rate and bit depth rather than a compression target, so this flag has minimal practical effect on the output — the true bitrate will be fixed by the audio's sample rate (e.g., ~1,411 kbps for 44.1 kHz stereo 16-bit). It is included here as a pass-through from the tool's default settings.
output.aifc Specifies the output filename with the .aifc extension, which tells FFmpeg to wrap the encoded pcm_s16be audio in the AIFC container format — Apple's extended AIFF format that supports both uncompressed and compressed audio codecs.

Common Use Cases

  • Importing legacy WMA music or voice recordings from old Windows Media Player libraries into Logic Pro, Pro Tools, or other professional DAWs on macOS that natively support AIFC/AIFF
  • Preparing WMA audio samples or sound effects for use in Apple's audio production ecosystem, where AIFC is a native and preferred format
  • Archiving WMA files from old Windows Media-based digital distribution platforms into a more open, platform-neutral PCM container for long-term preservation
  • Converting WMA voiceover recordings or podcast drafts into AIFC so they can be edited in professional audio editors like Audacity or Wavelab without codec compatibility issues
  • Migrating audio from Windows-centric workflows to Unix or macOS-based post-production pipelines that expect big-endian PCM audio in an AIFF-compatible container
  • Stripping Microsoft's proprietary codec dependency from audio files when handing off deliverables to studios or clients who require uncompressed PCM audio in a standard container

Frequently Asked Questions

No — converting from WMA to AIFC will not recover quality lost during the original WMA encoding. WMA uses lossy compression (wmav2), which permanently discards audio information during encoding. The conversion decodes the WMA to raw PCM and stores it in AIFC's lossless pcm_s16be format, so no further quality is lost in this step, but the AIFC file will sound identical to the WMA source, not better. Think of it as losslessly preserving whatever quality the WMA already contained.
WMA uses lossy compression to achieve small file sizes — a typical 128 kbps WMA file compresses audio significantly. The AIFC output uses uncompressed 16-bit PCM (pcm_s16be), which stores every audio sample without compression. A standard stereo 44.1 kHz pcm_s16be AIFC file runs at roughly 1,411 kbps, making it 10–20x larger than a 128k WMA file. This is expected and is the nature of lossless PCM audio containers.
Partial metadata preservation is possible, but results vary. WMA stores metadata using ASF (Advanced Systems Format) tags, while AIFC uses AIFF-style chunk-based metadata. FFmpeg will attempt to map common tags like title, artist, and album to their AIFC equivalents, but some WMA-specific or extended metadata fields may be dropped. If metadata fidelity is critical, verify the output file's tags using a tool like MediaInfo or your DAW after conversion.
AIFC extends AIFF by adding support for compressed audio codecs such as pcm_alaw, pcm_mulaw, and floating-point PCM variants (f32be, f64be), in addition to standard uncompressed PCM. In this conversion, the default output codec is pcm_s16be — standard 16-bit big-endian PCM — which is identical to what plain AIFF would store, making the output compatible with virtually any software that reads AIFF or AIFC files.
You can substitute the codec flag value to use a higher bit depth PCM format supported by AIFC. For example, replace '-c:a pcm_s16be' with '-c:a pcm_s24be' for 24-bit output or '-c:a pcm_f32be' for 32-bit floating-point, which is preferred by many professional DAWs. Note that since the source is a 128 kbps lossy WMA file, increasing bit depth will not recover lost audio detail — it just changes the numerical precision of the decoded samples stored in the output file.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.wma; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.wma}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.wma) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc"'. This applies the same pcm_s16be encoding to every WMA file in the directory and outputs a matching AIFC file for each one.

Technical Notes

The WMA format's default codec, wmav2, is a frequency-domain lossy codec that operates similarly to MP3 and AAC but is proprietary to Microsoft and bound to the ASF container. During decoding, FFmpeg uses its open-source wmav2 decoder to reconstruct PCM samples from the compressed bitstream. The output codec, pcm_s16be, stores 16-bit signed integers in big-endian byte order — the native byte order for AIFF/AIFC on classic Motorola and Apple hardware, which is why AIFC uses this format. One known limitation is that WMA files using DRM (Digital Rights Management) protection cannot be decoded by FFmpeg; the source file must be DRM-free for this conversion to succeed. WMA does not support multiple audio tracks, subtitles, or chapters, so none of those concerns apply here. The AIFC container similarly lacks subtitle and chapter support, making this a straightforward audio-only conversion with no stream selection complexity.

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