Extract Audio from 3G2 to AAC — Free Online Tool
Extract the AAC audio track from a 3G2 file and save it as a standalone .aac file. Since 3G2 files already use AAC as their default audio codec, this conversion is a direct stream extraction — no re-encoding required, preserving the original audio quality exactly as it was encoded for CDMA mobile networks.
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FFmpeg Command
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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
3G2 files store audio using AAC encoding by default, which is the same codec used in the .aac output container. Because the source codec and destination codec are identical, FFmpeg extracts the raw AAC audio bitstream directly from the 3G2 container and writes it to the .aac file without any re-encoding or transcoding. The video stream is discarded entirely using the -vn flag. This is essentially a demux operation rather than a transcode — the audio data itself is not touched, meaning there is zero additional quality loss beyond what was already present in the original 3G2 file. The result is a lightweight audio file stripped of all mobile-video container overhead.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser-based version of this tool, this runs via FFmpeg.wasm — a WebAssembly port of FFmpeg — so no files leave your device. |
-i input.3g2
|
Specifies the input file — a 3G2 container, a mobile multimedia format developed for CDMA networks that typically holds H.264 video and AAC audio in an ISO Base Media File Format structure. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 or MJPEG video stream in the 3G2 file. Since the output format is AAC (audio only), this flag ensures no video data is processed or written. |
-c:a aac
|
Sets the audio codec to AAC. Because the source 3G2 file already contains AAC audio, FFmpeg can perform a direct stream copy in practice; this flag confirms the target codec and ensures compatibility if the source audio ever differs. |
-b:a 128k
|
Sets the audio bitrate to 128 kilobits per second for cases where re-encoding occurs. For direct AAC-to-AAC extraction from 3G2, the original mobile-encoded bitrate is typically preserved, but this flag serves as the target if any transcoding step is triggered. |
output.aac
|
Defines the output filename with the .aac extension, directing FFmpeg to write a raw AAC audio bitstream file — a widely compatible audio format supported natively by Apple, Android, and most modern browsers and media players. |
Common Use Cases
- Recover a voice memo or audio clip recorded on an older CDMA phone (Verizon, Sprint) that saved footage as a 3G2 file, and convert it to a portable AAC file playable on modern devices.
- Extract the audio from a 3G2 video received via MMS or archived from an early 2000s–2010s CDMA handset to use in a podcast or audio project.
- Strip the audio from a 3G2 mobile video to reduce file size for archiving — keeping only the AAC audio when the video content is no longer needed.
- Prepare 3G2 audio content for iTunes or Apple ecosystem playback, since AAC is natively compatible with all Apple devices and apps.
- Extract a music clip or ringtone audio embedded in a 3G2 container to use as a standalone AAC audio file on a streaming or playback platform.
- Batch-process a collection of archived 3G2 mobile videos to extract their audio tracks for transcription, accessibility, or audio-only archiving purposes.
Frequently Asked Questions
No — because 3G2 files use AAC as their default audio codec, and the output format is also AAC, FFmpeg performs a direct stream copy rather than re-encoding. The audio bitstream is extracted byte-for-byte from the 3G2 container, so no additional compression or quality degradation occurs. Any quality limitations in the output reflect the original encoding decisions made when the 3G2 was first created.
3G2 was specifically designed for CDMA mobile networks with limited bandwidth, so audio tracks in 3G2 files are commonly encoded at low bitrates — often 64k or 96k — to minimize file size for over-the-air transmission. The extracted AAC file will reflect whatever bitrate the original 3G2 audio was encoded at. The -b:a 128k flag in the FFmpeg command applies only if re-encoding is triggered; in a pure stream extraction, the original bitrate is preserved.
Because this conversion extracts an already-encoded AAC stream without re-encoding, changing -b:a has no effect on the output quality — the original bitrate is preserved. If you want to re-encode at a higher bitrate (e.g., upscale to 192k), you can force re-encoding by explicitly adding -c:a aac -b:a 192k, but note that upsampling a low-bitrate mobile audio source does not restore lost quality. For most archival or playback purposes, direct extraction is the best approach.
The video stream is completely discarded. The -vn flag in the FFmpeg command instructs FFmpeg to ignore all video streams in the input 3G2 file and write only the audio to the output. Nothing about the video is preserved in the .aac file — AAC is a pure audio format with no support for video, subtitles, or chapters.
Significantly more compatible. The 3G2 container is a legacy mobile format tied to 3GPP2 and CDMA networks, and many modern media players, browsers, and platforms offer limited or no support for it. AAC, by contrast, is natively supported by Apple devices, Android, web browsers, streaming platforms, and virtually all modern audio software. Extracting the audio to .aac makes it universally accessible without needing a special codec or container parser.
On Linux or macOS, you can run a loop in the terminal: for f in *.3g2; do ffmpeg -i "$f" -vn -c:a aac -b:a 128k "${f%.3g2}.aac"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a aac -b:a 128k "%~nf.aac". This processes every 3G2 file in the current directory and outputs a matching .aac file for each one, making it efficient for archiving large collections of old mobile recordings.
Technical Notes
3G2 is a close relative of MP4 (both derive from the ISO Base Media File Format), and its default audio codec is AAC — the same codec used in the output container. This alignment means the conversion is effectively a demux: FFmpeg reads the AAC elementary stream from the 3G2 container and writes it directly into the .aac file without transcoding. One important caveat is that 3G2 files from CDMA-era devices often carry audio encoded at low sample rates (e.g., 22050 Hz or 8000 Hz for voice calls) and low bitrates due to the bandwidth constraints of CDMA networks. The extracted .aac file will inherit these characteristics. Metadata such as artist or title tags is rarely present in 3G2 files due to the format's mobile-first, minimal-overhead design, so the output .aac is likely to have sparse or empty metadata. The .aac raw bitstream format does not support chapters, multiple audio tracks, or embedded artwork — if those features are needed, consider targeting .m4a (an MPEG-4 audio container) instead, which wraps the same AAC codec with richer metadata support.