Convert 3G2 to WAV — Free Online Tool
Extract clean, uncompressed audio from 3G2 mobile video files by converting them to WAV format. This tool decodes the AAC audio track embedded in your 3G2 container and re-encodes it as 16-bit PCM — the standard uncompressed audio format — giving you a broadcast-ready WAV file with no lossy compression artifacts.
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 using AAC (Advanced Audio Coding), a lossy codec optimized for the low-bandwidth CDMA mobile networks these files were designed for. During this conversion, FFmpeg demuxes the 3G2 container to extract the AAC audio stream, fully decodes it to raw PCM samples, then re-encodes and writes those samples as 16-bit signed little-endian PCM inside a WAV container. There is no video output — WAV is audio-only. Because this involves a full decode-from-lossy step, the conversion is not lossless in an absolute sense: any compression artifacts introduced when the 3G2 was originally recorded are baked in. However, the resulting WAV will not introduce any additional generation loss, and it is fully uncompressed for downstream editing or archiving.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, it runs as FFmpeg.wasm compiled to WebAssembly — the same command works identically in a local terminal on Windows, macOS, or Linux for files over 1GB. |
-i input.3g2
|
Specifies the input file — a 3G2 container, which FFmpeg opens to demux its video and AAC audio streams. FFmpeg automatically detects the 3G2/MP4-family container format from the file header. |
-c:a pcm_s16le
|
Sets the audio codec to pcm_s16le — 16-bit signed little-endian PCM — which fully decodes the lossy AAC audio from the 3G2 file and writes raw, uncompressed samples into the WAV output. This is the standard uncompressed WAV encoding understood by every audio editor, DAW, and operating system. |
output.wav
|
Defines the output filename with a .wav extension. FFmpeg uses this extension to select the WAV muxer, which wraps the pcm_s16le audio stream in a standard RIFF/WAV container. The video stream from the 3G2 file is automatically dropped because WAV supports audio only. |
Common Use Cases
- Importing audio from old mobile phone video recordings into a DAW like Audacity or Adobe Audition, which works best with uncompressed WAV input
- Archiving voicemail or field recordings captured on early 2000s CDMA phones that saved files in 3G2 format before converting them to a future-proof format
- Extracting spoken-word audio from 3G2 clips to transcribe with speech-to-text software that requires PCM WAV input
- Preparing audio from 3G2 mobile videos for broadcast or podcast post-production workflows that mandate uncompressed source files
- Stripping the AAC audio from 3G2 videos to analyze waveform data or run audio forensics tools that need raw PCM samples
- Recovering the audio track from a 3G2 file when the video stream is corrupted or unplayable, salvaging the content as a standalone WAV
Frequently Asked Questions
No — and it is important to understand why. The AAC audio inside a 3G2 file was compressed at recording time, and that lossy compression is permanent. Converting to WAV decodes and preserves exactly what is in the 3G2 file without adding any further compression, so you will not lose additional quality in this step. However, WAV cannot recover detail that AAC already discarded, so the ceiling is whatever quality the original 3G2 recording had — typically low bitrate audio from a CDMA-era mobile device.
3G2 uses AAC audio, which achieves high compression ratios — often storing a minute of audio in under 1 MB. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression, which works out to roughly 10 MB per minute for stereo audio at 44.1 kHz. The size difference purely reflects the shift from heavily compressed mobile-optimized encoding to uncompressed archival storage — the audio content itself is unchanged.
Generally, no. 3G2 containers can carry metadata such as recording timestamps in their MP4-derived atom structure, but WAV's metadata support is limited to a basic INFO chunk. FFmpeg does attempt to map common tags, but location data and 3G2-specific atoms will be discarded. If preserving metadata is important, consider also keeping the original 3G2 file alongside the WAV.
Yes. The command uses pcm_s16le (16-bit signed little-endian PCM) by default, which is the most compatible WAV variant. You can substitute pcm_s24le or pcm_s32le for higher bit depths, or pcm_f32le for 32-bit float — for example: ffmpeg -i input.3g2 -c:a pcm_s24le output.wav. To change the sample rate, add -ar 44100 or -ar 48000 before the output filename. Since the original 3G2 source is low-bitrate AAC, going beyond 16-bit 44.1 kHz offers no real audible benefit but may be required by certain professional workflows.
On Linux or macOS, run: for f in *.3g2; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.3g2}.wav"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This iterates over every 3G2 file in the current directory and produces a matching WAV file. Batch processing on the desktop is especially practical for archiving large collections of old mobile recordings.
The video track is completely discarded. WAV is a pure audio container with no support for video streams, so FFmpeg automatically drops the video when writing to a WAV output file. Only the AAC audio stream is decoded and written as PCM audio. If you need to keep the video, consider exporting to a different format alongside this WAV extraction.
Technical Notes
3G2 is a close relative of MP4, built on the ISO Base Media File Format and designed specifically for CDMA2000 mobile networks (3GPP2). Its audio is almost universally AAC-LC at low bitrates (often 32–128 kbps), reflecting the bandwidth constraints of early mobile data. The FFmpeg command targets pcm_s16le — 16-bit signed PCM in little-endian byte order — which is the WAV codec used by virtually every piece of audio software and hardware, making it the safest default. WAV itself supports a range of codecs including FLAC (lossless compression), ADPCM, and A-law/mu-law variants, but pcm_s16le maximizes compatibility. One known limitation: 3G2 files sometimes contain multiple audio tracks or AMR-NB speech codec audio (common in very early 3GPP2 implementations); FFmpeg will select the default audio stream automatically, which may not always be the one you intend. Use -map 0:a:0 to explicitly select the first audio track if your file has multiple streams. Because WAV does not support chapters, subtitles, or multiple audio tracks, none of these — even if present in the 3G2 source — will survive the conversion.