Convert OGA to AAC — Free Online Tool
Convert OGA audio files (Ogg-wrapped Vorbis or FLAC) to AAC format using FFmpeg in your browser — no upload required. This conversion transcodes the audio stream from Vorbis or FLAC to AAC, making your files compatible with iTunes, iPhone, Apple Music, and virtually every mobile device and streaming platform.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your OGA file here
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Free — no uploads, no signups. Your files never leave your browser.
Settings
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 are Ogg container files carrying audio streams encoded with Vorbis, FLAC, or Opus. AAC is a fundamentally different codec with its own container format, so this conversion cannot be a simple remux — the audio stream must be fully decoded and re-encoded from scratch. FFmpeg decodes the OGA audio (stripping the Ogg container and decompressing the Vorbis or FLAC data to raw PCM audio), then re-encodes it using the native AAC encoder at 128k bitrate. Because OGA's lossless FLAC mode stores audio without degradation, converting FLAC-in-OGA to AAC introduces lossy compression for the first time. If the source was Vorbis-in-OGA (already lossy), this is a lossy-to-lossy transcode, and some generation loss occurs. Chapter metadata stored in OGA will be lost, as AAC does not support chapters.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool — in this browser-based tool, that means FFmpeg.wasm running entirely in WebAssembly inside your browser with no server involvement. On your desktop, this calls your locally installed FFmpeg binary. |
-i input.oga
|
Specifies the input file — an OGA audio container, which FFmpeg automatically detects and demuxes to extract the underlying audio stream (Vorbis, FLAC, or Opus depending on the source file). |
-c:a aac
|
Tells FFmpeg to encode the output audio using the native AAC encoder. This fully decodes the OGA audio stream (Vorbis or FLAC) to raw PCM and then re-encodes it as AAC, the codec required for iTunes, Apple Music, and iOS compatibility. |
-b:a 128k
|
Sets the target audio bitrate to 128 kilobits per second for the AAC output. At this bitrate AAC typically sounds transparent for most music and voice content, producing a file significantly smaller than a lossless FLAC-in-OGA source while maintaining good perceptual quality. |
output.aac
|
Defines the output filename and format. The .aac extension tells FFmpeg to write a raw AAC bitstream file rather than wrapping it in a container. Change this to output.m4a if you need a container with better metadata and seeking support. |
Common Use Cases
- Converting FLAC-quality audio distributed in OGA format into AAC files for syncing to an iPhone or iPad via iTunes or Apple Music
- Preparing OGA podcasts or audiobooks recorded in Vorbis for upload to Apple Podcasts, which requires AAC or MP3
- Making open-format OGA music files compatible with Android and iOS apps that lack native Ogg/Vorbis support
- Reducing file size of lossless FLAC-in-OGA audio for use in video projects or mobile apps where storage is constrained
- Batch-converting a Linux-generated OGA audio library to AAC so the files play natively in macOS Finder and QuickTime without plugins
- Transcoding OGA game audio or sound assets to AAC for use in iOS or macOS app bundles, which use AAC as the preferred audio format
Frequently Asked Questions
Yes — this is a lossy-to-lossy transcode. Vorbis audio inside the OGA file is already compressed, and decoding it then re-encoding it as AAC introduces a second generation of compression artifacts. At 128k AAC the result is generally transparent for typical listening, but it will never be bit-for-bit identical to the original. If audio fidelity is critical, always start from a lossless source such as FLAC-in-OGA rather than re-transcoding from Vorbis.
Yes. FLAC is lossless inside the OGA container, meaning the original waveform is preserved perfectly. However, AAC is a lossy format by design, so the re-encoding step introduces compression artifacts regardless of how pristine the source was. The upside is that starting from FLAC gives the AAC encoder the best possible input signal, which typically produces better-sounding output than transcoding from a previously lossy Vorbis source.
No. The Ogg container supports chapter metadata, and OGA files can carry chapter information, but the raw AAC container format (.aac) does not support chapters at all. If you need chapters preserved, consider converting to M4A or M4B instead of bare AAC, as those MPEG-4 containers support AAC audio along with chapter markers.
The FFmpeg command outputs a raw AAC bitstream file (.aac) rather than an MPEG-4 container (.m4a or .mp4). Both formats use AAC audio encoding, but M4A wraps the audio in an MPEG-4 container that adds metadata support, seeking efficiency, and broader compatibility with media players. If you need M4A instead, you can modify the FFmpeg command by changing the output filename to output.m4a — FFmpeg will automatically use the correct container.
Replace the value after -b:a with your desired bitrate. For example, use -b:a 256k for higher quality or -b:a 96k for a smaller file. AAC at 128k is considered transparent for most content, but voice recordings can sound fine at 96k, while high-quality music listening benefits from 192k or 256k. The full modified command would be: ffmpeg -i input.oga -c:a aac -b:a 256k output.aac
Yes, using a shell loop. On Linux or macOS, run: for f in *.oga; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.oga}.aac"; done — this processes every OGA file in the current directory and saves each as an AAC file with the same base name. On Windows Command Prompt, use: for %f in (*.oga) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac"
Technical Notes
OGA is a strict audio-only subset of the Ogg container and most commonly carries Vorbis audio encoded with the libvorbis encoder using a quality scale (VBR -q:a 0–10, default 4), though FLAC and Opus streams are also valid. AAC, by contrast, is a fixed-bitrate or VBR lossy codec standardized by MPEG and is the dominant format in the Apple ecosystem. FFmpeg's built-in AAC encoder (used here via -c:a aac) is a solid open-source implementation, though the optional libfdk_aac encoder (if compiled in) generally produces marginally better quality at the same bitrate. The -b:a 128k flag sets a target bitrate, not a hard cap, as FFmpeg's native AAC encoder uses a VBR-like mode internally. Metadata tags such as artist, title, and album that were embedded in the OGA file may not fully survive the container switch to raw AAC; wrapping in M4A preserves more metadata. The OGA's Ogg container overhead is also removed in the output, which slightly reduces file size independent of any re-encoding.