Convert OGA to AIFC — Free Online Tool
Convert OGA (Ogg Audio) files to AIFC format, transcoding Vorbis or FLAC streams into big-endian PCM audio compatible with professional Apple audio workflows. This tool runs entirely in your browser using FFmpeg.wasm — no uploads, no servers, free for files up to 1GB.
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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
OGA files store audio in the Ogg container, typically using the lossy Vorbis codec or lossless FLAC codec. AIFC is an extension of Apple's AIFF format that supports both uncompressed PCM and compressed audio. During this conversion, FFmpeg decodes the Ogg audio stream — whether Vorbis or FLAC — and re-encodes it as 16-bit big-endian signed PCM (pcm_s16be), which is the standard uncompressed format for AIFC. This means even if your source OGA used lossy Vorbis compression, the output AIFC will be uncompressed PCM, though the lossy artifacts from the original encoding are permanently baked in. If your source was FLAC-encoded OGA, the conversion to 16-bit PCM is lossless as long as the source bit depth does not exceed 16 bits.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the underlying engine that powers this browser-based tool via WebAssembly. The same command runs identically on a desktop FFmpeg installation for files over 1GB. |
-i input.oga
|
Specifies the input OGA file. FFmpeg will detect whether the Ogg container holds a Vorbis, FLAC, or Opus audio stream and select the appropriate decoder automatically. |
-c:a pcm_s16be
|
Sets the output audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio format for AIFC. This decodes the compressed Ogg audio (Vorbis, FLAC, or Opus) into raw waveform data stored in Apple's native big-endian byte order. |
-b:a 128k
|
Sets a target audio bitrate of 128 kbps. For PCM codecs like pcm_s16be, this flag has no practical effect because PCM bitrate is determined entirely by sample rate and bit depth rather than a compression target — it is included for command consistency but does not alter the output. |
output.aifc
|
Specifies the output filename with the .aifc extension, which causes FFmpeg to use the AIFF-C muxer and produce a file compatible with macOS audio applications, Logic Pro, Pro Tools, and other professional Apple-ecosystem tools. |
Common Use Cases
- Importing open-source game audio or music — often distributed as OGA files — into Logic Pro or GarageBand, which natively handle AIFC
- Preparing Vorbis-encoded audio assets for use in older Apple professional workflows that require AIFF-family formats
- Archiving FLAC-encoded OGA files to AIFC PCM for integration with legacy Mac-based digital audio workstations that predate Ogg support
- Converting Creative Commons or open-license music (commonly published in OGA) to AIFC for use in Final Cut Pro sound design timelines
- Supplying audio content to Apple ProRes or QuickTime-based pipelines that expect AIFF/AIFC audio rather than Ogg-wrapped streams
- Extracting and converting chapter-tagged OGA audiobook content to AIFC for editing in Pro Tools or other DAWs on macOS
Frequently Asked Questions
No. If your OGA file uses the Vorbis codec (the most common OGA codec), the audio has already been lossy-compressed and those artifacts are permanent. Converting to AIFC PCM simply decodes the Vorbis stream and stores the result uncompressed — it does not recover any detail that Vorbis discarded. You will get a larger, uncompressed file, but the audio fidelity is capped at whatever the original Vorbis encoding preserved.
FLAC is lossless, so decoding a FLAC-in-OGA file and writing it as 16-bit PCM AIFC is a lossless process — provided the FLAC source was recorded at 16-bit depth. If the original FLAC was 24-bit, the default pcm_s16be codec will truncate to 16 bits, which involves minor quantization. To preserve 24-bit depth, you would modify the FFmpeg command to use '-c:a pcm_s24be' instead.
OGA files use compression — Vorbis is lossy and typically achieves 8:1 to 12:1 compression ratios, while FLAC is lossless but still compresses by roughly 2:1. AIFC with pcm_s16be stores raw, uncompressed PCM audio with no compression at all. A 5-minute stereo track at 44.1 kHz will be roughly 50–60 MB as uncompressed 16-bit PCM, compared to perhaps 5–8 MB as Vorbis OGA.
Metadata tags such as artist, title, and album are typically carried over during conversion, though tag field naming conventions differ between Ogg and AIFC. However, AIFC does not support chapter markers — this is a fundamental limitation of the format, unlike OGA which does support chapters. Any chapter structure in your source file will be lost in the output AIFC.
Replace '-c:a pcm_s16be' with another supported PCM codec: 'pcm_s24be' for 24-bit, 'pcm_s32be' for 32-bit integer, 'pcm_f32be' for 32-bit float, or 'pcm_f64be' for 64-bit float. For example: 'ffmpeg -i input.oga -c:a pcm_s24be output.aifc'. The '-b:a 128k' bitrate flag is largely ignored by PCM codecs since PCM bitrate is determined by sample rate and bit depth, not a target bitrate setting.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.oga; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.oga}.aifc"; done'. On Windows PowerShell: 'Get-ChildItem *.oga | ForEach-Object { ffmpeg -i $_.FullName -c:a pcm_s16be -b:a 128k ($_.BaseName + ".aifc") }'. This is especially useful for batch-converting game audio asset libraries distributed in OGA format.
Technical Notes
The pcm_s16be codec produces signed 16-bit big-endian PCM, which is the native byte order for AIFC and compatible with the original Motorola 68k architecture that AIFF was designed for. Modern Intel and Apple Silicon Macs handle byte-order conversion transparently, so this has no practical compatibility impact. The '-b:a 128k' flag in the command has minimal effect on PCM codecs because PCM bitrate is a fixed function of sample rate and channel count — for a standard 44.1 kHz stereo file, 16-bit PCM always produces exactly 1,411.2 kbps regardless of the bitrate flag. Ogg metadata tags (stored as Vorbis comments) will be mapped to AIFF metadata fields where equivalents exist, but the field name mapping is not always one-to-one, and tags like 'TRACKNUMBER' may not survive intact. The libopus codec is also valid in OGA containers; if your source uses Opus, FFmpeg will decode it the same way and the conversion process is identical. AIFC does not support multiple audio tracks or subtitle streams, and neither does OGA's standard usage, so no stream data is lost in those respects beyond chapters.