Extract Audio from 3G2 to AU — Free Online Tool
Extract audio from 3G2 mobile video files and convert it to Sun AU format with 16-bit big-endian PCM encoding. This tool strips the AAC or MP3 audio track from CDMA-era mobile container files and outputs a lossless PCM waveform in the classic Unix AU format — useful for archival, signal processing, or legacy Unix/Java audio pipelines.
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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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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
The conversion works in two stages: first, FFmpeg discards the video stream entirely from the 3G2 container (which typically holds H.264 video alongside AAC audio). Second, the audio — originally compressed as AAC or MP3 inside the 3G2 file — is decoded and re-encoded as raw 16-bit big-endian PCM samples wrapped in a Sun AU container. Because AAC is a lossy codec, the output PCM is a faithful decoded representation of the compressed audio, not a perfect lossless reproduction of the original recording. The AU container itself adds only a minimal 24-byte header with sample rate, channel count, and encoding type metadata, making the resulting file significantly larger than the source 3G2 but immediately readable by any PCM-aware audio tool.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In the browser version this runs as FFmpeg.wasm compiled to WebAssembly, executing entirely client-side with no server upload. On the desktop you run the same command with a locally installed FFmpeg binary. |
-i input.3g2
|
Specifies the input file — a 3G2 container, the multimedia format developed for 3GPP2/CDMA mobile networks. FFmpeg will parse the ISO Base Media File Format structure of the 3G2 to locate the audio and video streams inside it. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 video stream in the 3G2 and produce an audio-only output. This is essential here because the AU format has no video support whatsoever. |
-c:a pcm_s16be
|
Decodes the AAC audio from the 3G2 and re-encodes it as 16-bit signed big-endian PCM — the standard encoding for the Sun AU format. Big-endian byte order matches the original Sun SPARC architecture specification for AU files. |
output.au
|
Specifies the output filename with the .au extension, which signals FFmpeg to wrap the PCM audio in a Sun AU container with the standard 24-byte header containing sample rate, channel count, and encoding type fields. |
Common Use Cases
- Recovering voice recordings or audio memos captured on older CDMA phones (Verizon, Sprint-era devices) that stored clips as 3G2 files, converting them into PCM AU for archival or forensic analysis
- Feeding 3G2 audio into legacy Unix-based audio processing pipelines or scientific signal analysis tools that expect Sun AU as their native input format
- Preparing audio extracted from 3G2 mobile clips for use in Java applications, which have built-in AU/PCM support through the javax.sound API without requiring additional codecs
- Decompressing AAC audio from 3G2 files into uncompressed PCM AU as a neutral intermediate format before re-encoding to a different target format with precise quality control
- Extracting audio from 3G2 video evidence files into a format suitable for audio waveform analysis or transcription workflows that require uncompressed PCM input
- Converting a collection of old 3G2 ringtones or audio clips from CDMA-era mobile devices into AU files for use in vintage Sun workstation emulators or retro computing projects
Frequently Asked Questions
The AU output will sound perceptually identical to the 3G2 audio, but it is not bit-for-bit lossless relative to the original recording. The 3G2 container stores audio as AAC or MP3, both of which are lossy codecs — meaning some audio information was discarded when the 3G2 was first created. This conversion decodes that compressed audio into uncompressed 16-bit PCM, so no additional quality is lost during the 3G2-to-AU step itself. Any quality reduction you notice was already present in the source file.
The 3G2 file stores audio as compressed AAC, which typically achieves 10:1 to 15:1 compression ratios over raw PCM. The AU file stores audio as uncompressed 16-bit big-endian PCM samples — every single audio sample is written directly to disk with no compression. For example, one minute of stereo audio at 44.1 kHz becomes roughly 10 MB as PCM, whereas the same audio might have been under 1 MB as AAC inside the 3G2. This size increase is expected and is a direct consequence of the decompression.
The output uses pcm_s16be, which means 16-bit signed PCM samples stored in big-endian byte order. This is the default and most widely supported encoding in the Sun AU format, matching the original Sun Microsystems specification. Big-endian byte order was standard on SPARC and Motorola 68k Unix workstations where AU originated. If you need a different encoding such as 8-bit PCM, mu-law (pcm_mulaw), or A-law (pcm_alaw), you can modify the -c:a flag in the FFmpeg command accordingly.
No. The Sun AU format has an extremely minimal header — it stores only the audio encoding type, sample rate, number of channels, and an optional free-form annotation field. It has no standardized support for ID3-style tags, creation timestamps, or the kind of metadata that modern mobile containers like 3G2 or MP4 can carry. Any title, artist, or date metadata embedded in the 3G2 file will be lost during this conversion.
You can add -ar followed by a sample rate (e.g., -ar 44100 or -ar 22050) and -ac followed by a channel count (e.g., -ac 1 for mono) before the output filename. For example: ffmpeg -i input.3g2 -vn -c:a pcm_s16be -ar 44100 -ac 1 output.au. This is useful if the source 3G2 has an unusual sample rate from a mobile device, and you need to normalize it for a specific downstream tool.
Yes. On Linux or macOS you can use a shell loop: for f in *.3g2; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.3g2}.au"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". This is particularly useful for batch-processing a collection of old mobile recordings, since the browser tool handles one file at a time while the desktop command handles the full set.
Technical Notes
The 3G2 container is structurally very close to MP4/ISO Base Media File Format, and its audio track is almost always AAC-LC at low bitrates (64–128 kbps) optimized for CDMA mobile transmission. When FFmpeg decodes this into pcm_s16be for the AU container, the decoded samples reflect the quantization artifacts inherent in the original AAC encoding — particularly audible as pre-ringing or HF rolloff if the source bitrate was low. The Sun AU format supports several encodings natively: pcm_s16be (the default here), pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw; the latter two are companded formats historically used in telephony and are smaller but lossy. Because AU files are headerless beyond the 24-byte preamble and contain no frame boundaries, they are trivially streamable and can be memory-mapped directly, making them popular in Java's javax.sound.sampled API and in legacy scientific audio toolchains. One known limitation: AU does not support multi-channel audio beyond stereo in most implementations, so if the 3G2 source somehow contains surround audio (rare but possible), only the first two channels will be preserved cleanly. Additionally, some 3G2 files from older CDMA devices encode audio at non-standard sample rates like 8000 Hz (narrowband voice quality) — the AU file will faithfully reproduce this rate, but you may want to resample with -ar 44100 if the destination application expects CD-quality audio.