Convert OGA to AIF — Free Online Tool
Convert OGA (Ogg audio with Vorbis, FLAC, or Opus streams) to AIF, Apple's uncompressed PCM audio format. This tool decodes your Ogg-encapsulated audio and re-encodes it as 16-bit big-endian PCM — the native lossless format for macOS audio workflows and professional audio software on Apple systems.
to
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 using compressed codecs — most commonly Vorbis (lossy) or FLAC (lossless) — inside an Ogg container. AIF uses raw, uncompressed PCM audio with no container compression. During this conversion, FFmpeg fully decodes the OGA audio stream to raw PCM samples in memory, then re-encodes them as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFF container. If your source is Vorbis or Opus, some generation loss is unavoidable since those are lossy codecs being decoded to lossless output — the AIF will be lossless, but it cannot recover detail discarded during the original lossy encoding. If your source is FLAC inside the OGA container, the decoded audio is mathematically lossless and the AIF output will be a true lossless representation of the original signal.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the Ogg container and its internal audio codec, and re-encoding the audio as uncompressed PCM in an AIFF container. |
-i input.oga
|
Specifies the input OGA file. FFmpeg automatically detects the Ogg container and identifies the internal audio codec (Vorbis, FLAC, or Opus) to determine how to decode the audio stream. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio encoding used in AIFF files. This fully decodes the compressed OGA audio and re-encodes it as raw PCM samples in the byte order required by the Apple AIFF specification. |
output.aif
|
Specifies the output filename with the .aif extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an AIFF container — Apple's lossless uncompressed audio format compatible with macOS audio tools, Logic Pro, GarageBand, and Final Cut Pro. |
Common Use Cases
- Importing open-source game audio or Linux-originated sound files (typically distributed as OGA with Vorbis) into Apple Logic Pro or GarageBand, which natively read AIF but not OGA
- Preparing OGA voice recordings or field recordings for use in Final Cut Pro timelines, where AIF is the preferred lossless interchange format
- Converting FLAC-in-OGA archival audio from a Linux or open-source pipeline into AIF for delivery to a mastering engineer using a Mac-based DAW
- Extracting and losslessly converting OGA sound effects downloaded from open-license audio libraries (like Freesound) into AIF for use in professional Mac audio post-production
- Migrating a music library stored in OGA format on Linux to an AIF-based workflow on macOS without introducing additional lossy encoding steps
- Converting OGA podcast source files into AIF for import into a Mac-based audio editor like Adobe Audition or Steinberg Nuendo running on macOS
Frequently Asked Questions
No — if your OGA file uses the Vorbis or Opus codec, those are lossy formats and any detail discarded during that original encoding is permanently gone. The AIF output will be uncompressed and lossless, but it is a lossless capture of already-lossy audio. You will not hear any improvement in quality; the output simply stops further generational loss by preserving what remains in full PCM fidelity. If your OGA contains FLAC audio, the conversion is fully lossless end-to-end.
OGA supports multiple codecs — Vorbis (lossy), Opus (lossy), and FLAC (lossless). The quality of your AIF output depends entirely on which codec was used inside the OGA container. A FLAC-in-OGA file converted to AIF yields a true lossless result. A Vorbis-in-OGA file yields an uncompressed copy of lossy audio — larger in file size than the source, but not higher in fidelity. You can identify the codec in your OGA file using a tool like MediaInfo before converting.
OGA stores audio using compressed codecs (Vorbis, FLAC, or Opus), which significantly reduce file size. AIF stores audio as raw, uncompressed PCM samples — every sample is stored at full bit depth with no compression whatsoever. A typical 5-minute stereo OGA file encoded in Vorbis might be around 5–10 MB, while the equivalent 16-bit 44.1kHz AIF will be roughly 50 MB. This size increase is expected and is the nature of uncompressed audio formats.
Basic metadata tags (such as artist, title, and album) may be partially preserved depending on how the AIF format supports them, but chapter markers will not transfer — AIF does not support chapters. OGA's Vorbis comment metadata fields do not map perfectly to AIFF's TEXT chunk metadata, so some tags may be lost or truncated during conversion. If metadata preservation is critical, review the output file in your target application after conversion.
Replace pcm_s16be in the command with a different PCM codec. For 24-bit AIF, use pcm_s24be: ffmpeg -i input.oga -c:a pcm_s24be output.aif. For 32-bit integer use pcm_s32be, and for 32-bit float use pcm_f32be. Higher bit depths produce larger files but can capture more dynamic range — this is most meaningful when your source is FLAC-in-OGA, since upsampling lossy Vorbis to 24-bit does not recover lost detail.
Yes — on macOS or Linux you can use a shell loop: for f in *.oga; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.oga}.aif"; done. On Windows Command Prompt, use: for %f in (*.oga) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". This processes every OGA file in the current directory and outputs a corresponding AIF file with the same base filename. The browser-based tool processes one file at a time, so the FFmpeg command approach is especially useful for bulk conversions.
Technical Notes
The conversion uses pcm_s16be — signed 16-bit big-endian PCM — as the output codec, which is the standard baseline for AIFF and matches CD audio bit depth (16-bit, typically at 44.1kHz). Big-endian byte ordering is a requirement of the AIFF specification, which originated on Motorola 68000-based Macs. FFmpeg will preserve the original sample rate from your OGA file unless you explicitly specify a different rate with -ar. OGA's Vorbis comment tags (TITLE, ARTIST, ALBUM, etc.) may be partially written to the AIFF metadata chunks, but the mapping is imperfect and application-dependent — do not rely on complete metadata fidelity without verification. OGA chapter data is silently dropped since AIFF has no chapter structure. If your OGA contains multiple audio streams (unusual but technically possible), only the first audio stream is mapped by default. For maximum fidelity from a FLAC-in-OGA source, consider using pcm_s24be or pcm_s32be to match or exceed the original FLAC bit depth.