Extract Audio from 3G2 to AIFF — Free Online Tool
Extract audio from a 3G2 mobile video file and save it as a lossless AIFF file, converting the compressed AAC audio track into uncompressed PCM (pcm_s16be) for full-fidelity playback on macOS and professional audio applications. This is ideal when you need to recover clean, archive-quality audio from CDMA-era mobile recordings.
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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
A 3G2 file typically carries AAC-encoded audio, which is a lossy compressed format optimized for low-bandwidth CDMA mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio, then re-encodes it into 16-bit big-endian PCM (pcm_s16be) — the default uncompressed codec used in AIFF files. Because the source audio is lossy AAC, there is a one-time decompression step: the AAC data is fully decoded and written as raw PCM samples. No further quality degradation occurs after this point, and the resulting AIFF file is a stable, uncompressed snapshot of whatever audio information survived the original mobile compression. The output file will be significantly larger than the 3G2 source because PCM stores every sample explicitly rather than using codec compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all decoding, stream selection, and encoding for this conversion. In the browser tool, this runs via FFmpeg.wasm (WebAssembly) without any server involvement. |
-i input.3g2
|
Specifies the input file — a 3G2 container, which is an MP4-family format developed for CDMA mobile networks typically carrying AAC audio and H.264 video at low bitrates. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 (or MJPEG) video stream in the 3G2 file. This is required because AIFF is an audio-only format and cannot contain video data. |
-c:a pcm_s16be
|
Sets the audio codec for the output to 16-bit signed big-endian PCM, which is the native uncompressed codec of the AIFF format. This decodes the lossy AAC audio from the 3G2 and writes it as raw, uncompressed PCM samples compatible with macOS and professional audio tools. |
output.aiff
|
Defines the output filename and format. The .aiff extension tells FFmpeg to wrap the pcm_s16be audio in an AIFF container — Apple's standard uncompressed audio format commonly used on macOS and in professional audio production environments. |
Common Use Cases
- Recovering audio from old 3G2 video clips recorded on early CDMA smartphones (e.g., Verizon or Sprint feature phones) for archiving on macOS in a lossless format
- Importing mobile-captured voice memos or field recordings stored as 3G2 into Logic Pro or GarageBand, which work natively with AIFF
- Extracting interview or dialogue audio from 3G2 news footage captured on CDMA devices for use in a professional audio editing workflow
- Preparing 3G2 audio content for mastering or audio restoration software that requires uncompressed PCM input rather than compressed AAC
- Converting a 3G2 ringtone or audio clip from an old mobile backup into an AIFF file compatible with macOS system audio tools or iTunes/Music app
- Stripping and losslessly storing the audio from a 3G2 video to free up space while retaining maximum audio fidelity without re-compression
Frequently Asked Questions
No — converting 3G2 to AIFF does not recover quality lost during the original AAC encoding inside the 3G2 file. AAC is a lossy codec, so some audio detail was permanently discarded when the 3G2 was first created, typically at a low bitrate suited for CDMA mobile networks. What AIFF gives you is a lossless container that faithfully stores all remaining audio data without adding any further compression artifacts. Think of it as locking in the current quality, not restoring lost quality.
3G2 files store audio using AAC compression, which achieves very small file sizes (often 64–128 kbps) specifically for low-bandwidth mobile transmission. AIFF with pcm_s16be stores every audio sample as a raw 16-bit value — typically producing files around 10 MB per minute of stereo audio at 44.1 kHz. Since PCM applies zero compression, the size difference between a 3G2 and its AIFF equivalent can easily be 10–20x. This is expected and is the nature of converting from compressed to uncompressed audio.
The output uses pcm_s16be — 16-bit signed big-endian PCM — which is the standard default codec for AIFF files. The 'big-endian' byte order is native to the AIFF format, which Apple designed with big-endian architecture in mind (originally for 68k and PowerPC Macs). A 16-bit depth is sufficient for most audio sourced from 3G2 files, which were originally encoded at low AAC bitrates and would not benefit meaningfully from 24-bit or 32-bit depth.
Yes. You can replace pcm_s16be with another supported AIFF codec such as pcm_s24be (24-bit) or pcm_s32be (32-bit) by modifying the -c:a flag: ffmpeg -i input.3g2 -vn -c:a pcm_s24be output.aiff. However, given that the source is compressed AAC audio from a mobile 3G2 file, increasing the bit depth beyond 16-bit will not recover detail that AAC discarded — it will only produce a larger file. A higher bit depth is more useful when your source is already high-resolution lossless audio.
Metadata support varies. 3G2 files can carry limited metadata in their MP4-style container (such as title or creation date), and FFmpeg will attempt to map compatible tags into the AIFF output. However, AIFF has limited metadata support compared to formats like FLAC or MP4, so not all fields may transfer. GPS data, mobile device identifiers, or 3GPP2-specific metadata embedded in the 3G2 container will generally not carry over to AIFF.
Yes, on macOS or Linux you can use a shell loop: for f in *.3g2; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.3g2}.aiff"; done. On Windows Command Prompt you can use: for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". This processes each 3G2 file in the current directory and writes a corresponding AIFF file with the same base name. The browser-based tool handles one file at a time, so the FFmpeg command is especially valuable for bulk workflows.
Technical Notes
3G2 is a variant of the MP4/ISOBMFF container family developed for 3GPP2 (CDMA network) mobile devices, and its audio is almost universally AAC at low bitrates (often 64–96 kbps, mono or narrow stereo), reflecting the bandwidth constraints of CDMA data channels. When FFmpeg decodes this AAC stream and writes it as pcm_s16be AIFF, the sample rate of the output matches whatever rate was used in the 3G2 — commonly 8 kHz or 16 kHz for voice-focused recordings, or 44.1 kHz for media files. Users should be aware that 8 kHz telephone-quality audio will remain 8 kHz in the AIFF; the format conversion does not upsample. The -vn flag ensures no video data is written, which is necessary because AIFF is a pure audio format with no video container capability. AIFF does not support chapters, subtitles, or multiple audio tracks, but 3G2 files rarely carry more than one audio track anyway. The resulting AIFF is fully compatible with macOS Finder previews, Logic Pro, GarageBand, Pro Tools, and any application that reads standard Apple audio files.