Convert 3G2 to AAC — Free Online Tool
Extract and convert the AAC audio track from a 3G2 mobile video file into a standalone AAC audio file. Since 3G2 files already use AAC as their default audio codec, this conversion is essentially a stream extraction — the audio data is re-encoded at your chosen bitrate with minimal quality degradation.
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
A 3G2 file is a multimedia container holding both video and audio streams, where the audio is almost always encoded as AAC — the same codec used in the output. During this conversion, FFmpeg discards the video stream entirely and re-encodes (or in practice very nearly passes through) the AAC audio track into a raw AAC file at the target bitrate of 128k. Because no video codec processing is involved, this is a fast, CPU-light operation. The output is a pure .aac audio file with no container overhead, suitable for direct playback, streaming, or further processing.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.3g2
|
Specifies the input file — a 3G2 multimedia container from the 3GPP2 mobile standard, which typically holds a low-bitrate H.264 video stream and an AAC audio stream recorded on a CDMA mobile device. |
-c:a aac
|
Sets the audio codec to AAC using FFmpeg's built-in AAC encoder. Since the source 3G2 file almost certainly already contains AAC audio, this re-encodes the stream into a clean AAC output at the specified bitrate. |
-b:a 128k
|
Sets the output audio bitrate to 128 kilobits per second. This is higher than the typical 3G2 source bitrate of 24k–64k, so it ensures the output isn't artificially compressed further — though it cannot restore quality that the original mobile recording discarded. |
output.aac
|
The output filename with a .aac extension, telling FFmpeg to write a raw AAC bitstream (ADTS format) with no surrounding container — the simplest and most universally playable form of AAC audio, compatible with Apple devices, browsers, and most media players. |
Common Use Cases
- Extracting a voicemail or voice memo recorded on an older CDMA phone (e.g., Verizon or Sprint) that saved files in 3G2 format, so it can be played back on modern devices or transcribed
- Pulling the audio from a 3G2 video clip to create a ringtone or notification sound compatible with iTunes and iOS devices
- Archiving the audio commentary or narration from a 3G2 mobile video without carrying the large, low-quality video payload
- Preparing audio from legacy 3G2 footage for use in a podcast or audio edit, stripping the degraded mobile video to work with just the sound
- Converting 3G2 audio recordings from early-2000s CDMA-era camcorders or phones into AAC files for upload to music or podcast platforms that don't accept video containers
- Reducing file size by extracting only the audio from a 3G2 clip before sharing it in a chat app or email where video attachments are blocked
Frequently Asked Questions
Because 3G2 files already store audio as AAC, you are technically re-encoding from one AAC stream to another, which introduces a small generational quality loss. However, 3G2 audio is typically recorded at very low bitrates (32k–64k) due to CDMA network constraints, so encoding the output at 128k will not recover quality that was never there — but it also won't meaningfully degrade what exists. If you want to minimize loss, match the output bitrate to the original source bitrate rather than upsampling.
A 3G2 file bundles both a video stream and an audio stream into a single container. When you extract just the audio, you discard the video data entirely, which typically accounts for the vast majority of the file size. A 10MB 3G2 video might produce an AAC audio file of only 500KB–1MB depending on the clip's duration and the output bitrate, because mobile-recorded AAC audio is compact by design.
3G2 containers can store limited metadata, but raw .aac files have no standardized metadata container — they are a bare audio bitstream without ID3 tags or equivalent. Any metadata embedded in the 3G2 file will not be carried over by this FFmpeg command. If metadata preservation matters, consider converting to M4A instead, which wraps AAC in an MPEG-4 container that supports full tagging.
Yes — AAC is Apple's preferred audio codec and is natively supported across all iOS devices, macOS, and iTunes/Apple Music. A raw .aac file will play directly in QuickTime Player and will import into iTunes without any additional plugins or conversion steps, making this one of the most straightforward ways to get mobile audio onto Apple platforms.
Replace the value after '-b:a' with your desired bitrate. For example, use '-b:a 64k' to match a typical 3G2 source bitrate and avoid upsampling, or '-b:a 192k' for higher quality output if the source audio warrants it. Supported values range from 64k up to 320k. Note that increasing the bitrate beyond what the 3G2 source captured won't recover lost audio detail — it will only increase file size.
Yes, on the command line you can use a shell loop to process multiple files. On Linux or macOS, run: for f in *.3g2; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.3g2}.aac"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". The browser-based tool processes one file at a time, so the command line approach is recommended for batch jobs.
Technical Notes
3G2 (3GPP2) was designed for CDMA networks like those operated by Verizon and Sprint, and its audio tracks are almost universally encoded as AAC-LC at low bitrates — commonly 24k to 64k — to minimize data transmission costs. This means the source audio quality is already significantly constrained before conversion begins. The output format is a raw AAC bitstream (ADTS framing), which is the most universally compatible form of AAC but lacks a metadata container. If you need chapter markers, album art, or ID3-style tags, the output format should instead be M4A (AAC wrapped in MPEG-4). The 3G2 container also supports AMR-NB as an audio codec in some older files — if your 3G2 file uses AMR audio rather than AAC, FFmpeg will still transcode it, but the '-c:a aac' flag ensures the output is always standard AAC regardless of the source codec. No subtitle or chapter data exists in 3G2 to be lost in this conversion. The special '-movflags +faststart' flag used internally for 3G2 input handling is irrelevant to the AAC output, which has no container structure requiring optimization.