Convert 3G2 to AIF — Free Online Tool
Convert 3G2 mobile video files to AIF, extracting the compressed AAC audio track and re-encoding it as uncompressed PCM audio for high-fidelity playback on Mac systems. This tool strips the CDMA-era video container entirely and delivers a lossless AIF file ready for use in professional audio workflows.
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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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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
3G2 files typically carry AAC-encoded audio, which is a lossy compressed format designed for low-bitrate transmission over CDMA mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio, then re-encodes it to PCM signed 16-bit big-endian (pcm_s16be) — the uncompressed pulse-code modulation format that AIF uses. This decode-then-re-encode step means the audio is decompressed from its lossy AAC state into a fully uncompressed waveform. The resulting AIF file is significantly larger than the source 3G2 but contains no further compression artifacts and is natively readable by macOS apps like Logic Pro, GarageBand, and QuickTime.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In this browser-based tool, FFmpeg runs entirely via WebAssembly (FFmpeg.wasm) inside your browser — no server receives your file. On your desktop, this calls your locally installed FFmpeg executable. |
-i input.3g2
|
Specifies the input file in 3G2 format. FFmpeg reads the 3GPP2 container, identifying the contained video stream (typically H.264) and audio stream (typically AAC) before processing begins. |
-c:a pcm_s16be
|
Decodes the AAC audio from the 3G2 container and re-encodes it as PCM signed 16-bit big-endian — the uncompressed audio codec required by the AIF/AIFF specification. Big-endian byte order is mandatory here because the AIFF format was designed around Motorola/PowerPC architecture conventions. |
output.aif
|
Defines the output filename and tells FFmpeg to write an AIF (Audio Interchange File Format) container. FFmpeg infers the AIF container from the '.aif' extension, and since AIF is an audio-only format, the video stream from the 3G2 source is automatically discarded. |
Common Use Cases
- Recovering audio from an old 3G2 video recorded on a CDMA phone (e.g., Verizon or Sprint) to import into a Mac-based DAW like Logic Pro for editing or mixing
- Archiving voice memos or field recordings captured in 3G2 format to uncompressed AIF for long-term preservation without additional lossy compression cycles
- Extracting a music performance or rehearsal clip recorded on an older 3G2-capable device into AIF so it can be used as a sample in GarageBand or Ableton
- Preparing audio from a 3G2 source file for broadcast or podcast post-production workflows that require uncompressed input formats
- Converting a 3G2 ringtone or audio clip for use in Final Cut Pro or other Apple professional tools that prefer AIF over compressed mobile formats
- Stripping and uncompressing audio from legacy 3G2 archives before batch-mastering them to a final delivery format
Frequently Asked Questions
No — converting from 3G2 to AIF will not recover quality that was lost when the original AAC audio was encoded. AAC is a lossy format, so some audio detail was discarded at the time of recording. What AIF gives you is an uncompressed representation of whatever quality the AAC preserved, ensuring no further degradation occurs in subsequent editing or processing steps. Think of it as freezing the quality in place, not restoring it.
3G2 files use AAC audio, which typically achieves compression ratios of 10:1 or higher compared to uncompressed audio. AIF stores audio as raw PCM samples with no compression at all — at CD quality (16-bit, 44.1kHz stereo), that's about 10MB per minute. A 3G2 clip that was only 1MB due to AAC compression at 128k bitrate could expand to 10MB or more as a pcm_s16be AIF file. This is expected behavior for a lossy-to-lossless conversion.
3G2 files can carry basic metadata like title or artist tags, but the AIF format has limited metadata support compared to containers like MP4 or FLAC. FFmpeg will attempt to copy compatible metadata fields during conversion, but some 3G2-specific tags (especially those tied to 3GPP2 mobile standards) may not map cleanly to AIF's ID3-style metadata chunks. For critical metadata preservation, review the output file in a tool like MediaInfo after conversion.
The video stream is completely dropped. The FFmpeg command targets only the audio stream with '-c:a pcm_s16be' and writes to an AIF output, which is an audio-only format. FFmpeg automatically omits any streams that have no corresponding output container support — since AIF cannot hold video, the video data is discarded without any explicit flag needed.
You can replace 'pcm_s16be' with an alternative PCM codec to change the bit depth. For 24-bit audio use 'pcm_s24be', for 32-bit integer use 'pcm_s32be', and for 32-bit or 64-bit float use 'pcm_f32be' or 'pcm_f64be' respectively. For example: 'ffmpeg -i input.3g2 -c:a pcm_s24be output.aif'. Higher bit depths increase file size further but are preferred in professional mastering environments. Note that since the source AAC audio in the 3G2 rarely exceeds 16-bit equivalent resolution, 'pcm_s16be' is sufficient for most use cases.
Yes. On macOS or Linux, you can use a shell loop: 'for f in *.3g2; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.3g2}.aif"; done'. On Windows Command Prompt, use: 'for %f in (*.3g2) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif"'. This is especially useful for archiving a folder of old CDMA phone recordings in bulk. The browser-based tool handles one file at a time, so the FFmpeg command is the recommended approach for large batches.
Technical Notes
3G2 was designed for constrained CDMA network transmission, so its audio streams are almost universally AAC at modest bitrates (64k–128k is common), and the container is a close relative of MP4/MOV with a different brand identifier. The FFmpeg command for this conversion performs a full audio transcode: AAC is fully decoded to PCM in memory, then written as big-endian 16-bit PCM into an AIF/AIFF container — big-endian byte order is a requirement of the AIFF specification, which is why 'pcm_s16be' rather than 'pcm_s16le' is used. The AIF format does not support subtitles, chapters, or multiple audio tracks, and since 3G2 itself rarely contains these features, no data loss beyond the video stream is expected. One known limitation is that 3G2 files from certain older handsets may use AMR-NB or EVRC audio codecs instead of AAC; FFmpeg handles these automatically during decode, but the resulting audio quality will reflect the narrowband telephone-grade source (typically 8kHz sample rate), which will be accurately preserved in the uncompressed AIF output.