Extract Audio from 3G2 to AIF — Free Online Tool
Extract audio from 3G2 mobile video files and save it as a high-quality, uncompressed AIF file. This tool decodes the AAC audio stream from the 3G2 container and re-encodes it to 16-bit big-endian PCM (pcm_s16be), Apple's native lossless format — perfect for bringing mobile-captured audio into professional Mac-based workflows.
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
3G2 files store audio as compressed AAC, which is a lossy codec optimized for low-bandwidth CDMA mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio to raw PCM samples, then writes them into an AIF container using the pcm_s16be codec — 16-bit signed big-endian PCM. Because AAC is lossy, the decoded audio reflects whatever quality was captured on the mobile device; the AIF output is a lossless, uncompressed representation of that decoded audio. There is no re-compression step, so no further quality is lost beyond what the original 3G2 encoding already introduced. The resulting AIF file will be significantly larger than the source 3G2 due to the uncompressed nature of PCM audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no files leave your device. |
-i input.3g2
|
Specifies the input file — a 3G2 container, which typically holds an H.264 video stream and an AAC audio stream encoded for CDMA mobile network transmission. |
-vn
|
Disables video output entirely, discarding the H.264 video stream from the 3G2 file. This is required because AIF is a pure audio container and cannot hold a video stream. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding for AIF files. This decodes the compressed AAC audio from the 3G2 source into raw PCM samples, producing a lossless AIF file that introduces no further compression artifacts. |
output.aif
|
Defines the output filename with the .aif extension. FFmpeg uses this extension to determine that the output container should be Apple's Audio Interchange File Format, which is natively supported by macOS applications like Logic Pro, GarageBand, and QuickTime. |
Common Use Cases
- Importing voice memos or field recordings captured on older CDMA mobile phones into Logic Pro or GarageBand, which work natively with AIF files
- Recovering spoken audio from 3G2 video clips recorded on legacy Verizon or Sprint devices for transcription or archival purposes
- Extracting audio from 3G2 footage shot on early smartphones to use as stems in a Mac-based audio production session
- Preparing mobile-recorded audio evidence or reference recordings in an uncompressed, archival-grade format for legal or journalistic use
- Converting 3G2 voiceovers or on-location audio grabs into AIF so they can be imported without transcoding issues into Final Cut Pro timelines
- Stripping audio from 3G2 video files received via MMS or older multimedia messaging systems to preserve the audio track in a non-proprietary lossless format
Frequently Asked Questions
No — the AIF output will not sound better than the original 3G2 audio. The AAC audio inside a 3G2 file is already lossy, meaning some audio information was permanently discarded when the mobile device recorded or encoded it. Converting to AIF creates a lossless, uncompressed copy of the decoded AAC audio, which prevents any further quality loss, but it cannot restore what AAC compression already removed. Think of it as freezing the quality at its current level in a lossless container.
3G2 files use AAC audio, which is a highly compressed lossy codec designed for mobile networks with limited bandwidth — typical AAC bitrates in 3G2 files are 64–128 kbps. AIF with pcm_s16be stores raw, uncompressed PCM audio, which for stereo audio at a standard 44.1 kHz sample rate consumes roughly 1.4 MB per minute of audio per channel. A 3G2 file that is a few hundred kilobytes can expand into an AIF file several times that size, which is expected and correct behavior.
Yes — FFmpeg preserves the original sample rate and channel count from the 3G2 file's AAC stream when writing to AIF. If the 3G2 was recorded in mono at 8 kHz (common for voice calls on CDMA networks), the AIF will also be mono at 8 kHz. If you need a different sample rate, you can add the '-ar' flag to the FFmpeg command, for example '-ar 44100' to upsample to 44.1 kHz, though upsampling cannot recover frequency content that AAC or the original recording discarded.
Yes — AIF supports several PCM codecs beyond pcm_s16be. You can replace '-c:a pcm_s16be' in the FFmpeg command with '-c:a pcm_s24be' for 24-bit or '-c:a pcm_s32be' for 32-bit output. For most 3G2 source material recorded on mobile devices, 16-bit is more than sufficient since the original AAC audio rarely captures dynamic range that would benefit from higher bit depth, but 24-bit output is useful if the AIF will be further processed in a DAW that operates at 24-bit internally.
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}.aif"; done'. On Windows Command Prompt, use: 'for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif"'. This iterates over all 3G2 files in the current directory and produces a matching AIF file for each, which is practical when processing a batch of archived mobile recordings.
3G2 files can carry limited metadata in their MPEG-4 container structure, but AIF has a minimal metadata specification and FFmpeg's support for mapping 3G2 metadata tags to AIF ID3 or MARK chunks is limited. In practice, most 3G2 files from mobile devices carry little to no meaningful metadata beyond technical stream parameters, so metadata loss is rarely a concern for this specific conversion. If metadata preservation is critical, inspect the source file with 'ffprobe input.3g2' first to see what tags are present.
Technical Notes
The 3G2 container is essentially a restricted profile of MPEG-4 Part 12, optimized for 3GPP2 (CDMA) mobile networks, and its audio track is almost universally AAC — often at low bitrates (64–128 kbps) and sometimes at reduced sample rates like 8 kHz or 16 kHz for voice-oriented recordings. The '-vn' flag is essential here because 3G2 files contain a video stream (typically H.264) that must be explicitly suppressed; without it, FFmpeg would attempt to transcode the video into AIF, which is an audio-only container and would cause an error. The pcm_s16be codec is the standard default for AIF, matching Apple's historical use of big-endian byte ordering in the format. One limitation to be aware of: if the source 3G2 file has a very low sample rate (e.g., 8 kHz, typical of CDMA voice recordings), the resulting AIF will sound narrow and telephone-quality — this is a property of the source material, not a conversion artifact. AIF does not support multiple audio tracks, subtitles, or chapters, but 3G2 files from mobile devices virtually never contain these either, so no data loss occurs beyond the intentional video stream removal.