Extract Audio from 3G2 to CAF — Free Online Tool
Extract audio from 3G2 mobile video files and save it as a CAF file using Apple's Core Audio Format container. This tool decodes the AAC audio track from your 3G2 file and re-encodes it as uncompressed PCM (pcm_s16le) inside CAF — ideal for importing into Logic Pro, GarageBand, or other Apple audio tools that expect high-fidelity source material.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your 3G2 file here
or click to browse
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
3G2 files typically carry AAC-encoded audio alongside H.264 video, packaged in a container designed for CDMA mobile networks. This conversion strips out the video stream entirely and decodes the compressed AAC audio, then re-encodes it as 16-bit signed little-endian PCM (pcm_s16le) wrapped in Apple's CAF container. Unlike formats such as AIFF or WAV, CAF has no 4GB file size ceiling, making it a robust choice for long recordings. Because the audio is being decoded from lossy AAC and re-encoded as uncompressed PCM, the output file will be significantly larger than the source — but no additional quality is lost beyond what AAC compression already removed. The video stream is discarded with the -vn flag and never processed.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, FFmpeg runs via WebAssembly (FFmpeg.wasm) entirely client-side — your 3G2 file is never uploaded to a server. |
-i input.3g2
|
Specifies the input file — a 3G2 container, which typically holds H.264 video and AAC audio encoded for CDMA mobile network transmission. FFmpeg reads both streams but will only process the audio based on the flags that follow. |
-vn
|
Disables video output entirely. This tells FFmpeg to skip the H.264 (or MJPEG) video stream in the 3G2 file and produce an audio-only output, which is required since CAF is a purely audio container. |
-c:a pcm_s16le
|
Sets the audio codec to 16-bit signed little-endian PCM — uncompressed audio. This decodes the lossy AAC audio from the 3G2 source and stores it as raw PCM inside the CAF container, making it immediately compatible with Apple's Core Audio framework and DAWs like Logic Pro. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For PCM output (pcm_s16le), this flag has no practical effect since uncompressed PCM bitrate is fixed by sample rate and bit depth — it becomes relevant if you switch to a compressed codec like AAC or libopus for the CAF output. |
output.caf
|
Defines the output file name and tells FFmpeg to use the CAF (Core Audio Format) container. FFmpeg infers the container from the .caf extension, wrapping the pcm_s16le audio data in Apple's chunk-based CAF structure, which supports files larger than 4GB unlike WAV or AIFF. |
Common Use Cases
- Recovering audio from old 3G2 videos recorded on early CDMA phones (e.g., Verizon or Sprint era devices) for archiving in Apple's ecosystem
- Importing a 3G2 voice memo or field recording into Logic Pro or GarageBand, which natively support CAF as a project audio format
- Preparing extracted mobile video audio as a clean PCM source for noise reduction or audio restoration workflows in macOS tools like RX by iZotope
- Extracting narration or commentary audio from a 3G2 video file to use as a voiceover track in Final Cut Pro
- Converting 3G2 audio content to uncompressed CAF for lossless round-tripping in a mastering chain without introducing further codec artifacts
- Archiving audio from 3G2 video files sent via MMS or downloaded from legacy mobile platforms into an uncompressed, future-proof format
Frequently Asked Questions
No — the output will not be higher quality than the source. The 3G2 file stores audio as compressed AAC, which is a lossy codec. This conversion decodes that AAC audio and stores it as uncompressed PCM in the CAF file, which is lossless from this point forward, but any quality loss from the original AAC encoding is already baked in. What you gain is a format that is natively compatible with Apple's audio tools and free from further compression artifacts during editing.
The 3G2 file uses AAC, a lossy compression codec that typically encodes audio at 64–128 kbps. The output CAF file uses pcm_s16le — uncompressed 16-bit PCM — which for stereo audio at 44.1 kHz runs around 1.4 Mbps, roughly 10–20 times larger. This is expected behavior and the reason CAF was designed to support large file sizes without the 4GB limit that WAV and AIFF impose.
CAF is Apple's proprietary container format and is not natively supported on Windows or Linux. While the pcm_s16le audio data inside is universal, most non-Apple media players and DAWs will not open CAF directly. If you need cross-platform compatibility, consider converting to WAV or FLAC instead. CAF is best suited for workflows that stay within macOS or iOS audio tools like Logic Pro, GarageBand, or Core Audio APIs.
Generally no. 3G2 uses a metadata structure based on the MP4/ISOBMFF specification, while CAF uses Apple's own chunk-based metadata format. FFmpeg does not automatically map 3G2 metadata tags into CAF-compatible chunks during this conversion. If preserving metadata is important, you would need to add -metadata flags to the command or edit the CAF file post-conversion using a tool like afinfo or a DAW.
To use AAC instead of uncompressed PCM in the CAF output, replace '-c:a pcm_s16le' with '-c:a aac' and adjust '-b:a 128k' to your preferred bitrate (e.g., 192k or 256k). CAF also supports FLAC for lossless compressed output — use '-c:a flac' and remove the -b:a flag, since FLAC uses a compression level rather than a bitrate. For example: ffmpeg -i input.3g2 -vn -c:a flac output.caf gives you lossless compression in a much smaller file than PCM.
Yes. On macOS or Linux, you can use a shell loop: for f in *.3g2; do ffmpeg -i "$f" -vn -c:a pcm_s16le -b:a 128k "${f%.3g2}.caf"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a pcm_s16le -b:a 128k "%~nf.caf". This processes each 3G2 file in the current directory and outputs a corresponding CAF file with the same base filename.
Technical Notes
The 3G2 container is a direct descendant of the MPEG-4 Part 12 specification, meaning its audio packaging is structurally similar to MP4 and M4A. However, FFmpeg cannot simply remux the AAC stream from 3G2 into CAF without re-encoding, because the AAC bitstream encapsulation differs between the two containers (3G2 uses MPEG-4 AudioSpecificConfig framing while CAF wraps audio in its own chunk structure). The result is a full decode-and-encode cycle through pcm_s16le. The -b:a 128k flag is included in the command but has no effect on PCM output — PCM bitrate is determined entirely by sample rate and bit depth, not a target bitrate. If you switch to a codec like AAC or libopus inside CAF, the -b:a flag becomes meaningful. Note that CAF's pcm_s16le output will truncate any 24-bit or 32-bit audio in the source to 16-bit; if the 3G2 source contains higher-bit-depth audio (uncommon but possible), switch to pcm_s24le or pcm_s32le to preserve full fidelity. Also note that 3G2 does not support multiple audio tracks, so there is no risk of accidentally discarding a secondary audio stream.