Extract Audio from 3G2 to WAV — Free Online Tool
Extract audio from 3G2 mobile video files and save it as uncompressed WAV audio. This tool decodes the AAC audio stream typically found in 3G2 files and outputs it as 16-bit PCM WAV — converting from a lossy compressed mobile format to a lossless, broadcast-ready audio file.
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 low-bitrate transmission over CDMA mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio, then re-encodes it as PCM signed 16-bit little-endian (pcm_s16le) — the standard uncompressed audio codec used in WAV files. Because AAC is lossy, the original compression artifacts cannot be undone, but the output WAV will be a fully uncompressed, artifact-free representation of whatever audio quality existed in the 3G2 source. This makes it ideal for further editing or archiving without introducing additional generation loss.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly — the same command works identically on desktop FFmpeg installations for files over 1GB. |
-i input.3g2
|
Specifies the input 3G2 file. FFmpeg reads the MPEG-4-derived container and identifies all streams — typically one H.264 video stream and one AAC audio stream conforming to the 3GPP2 mobile standard. |
-vn
|
Disables video output entirely. Since WAV is an audio-only container, this flag tells FFmpeg to ignore the H.264 (or other) video stream in the 3G2 file rather than attempting to write it, which would cause an error. |
-c:a pcm_s16le
|
Decodes the AAC audio from the 3G2 source and re-encodes it as PCM signed 16-bit little-endian — the standard uncompressed audio codec for WAV files, offering universal compatibility with audio editors, DAWs, and broadcast systems. |
output.wav
|
Defines the output file as a WAV container. FFmpeg uses the .wav extension to select the RIFF/WAV muxer, which wraps the uncompressed pcm_s16le audio stream in the standard Waveform Audio File Format structure. |
Common Use Cases
- Recovering audio from old 3G2 video clips recorded on early CDMA phones (Verizon, Sprint) for archiving into a lossless format
- Extracting speech or dialogue from 3G2 mobile video to transcribe or process with audio editing software that doesn't support the 3G2 container
- Preparing mobile-recorded audio for use in broadcast or professional video production workflows that require uncompressed WAV sources
- Stripping background music or ambient audio from 3G2 footage to use as a standalone sound effect or audio sample
- Converting 3G2 voicememos or call recordings from legacy CDMA devices into WAV for long-term storage and compatibility
- Feeding extracted audio into digital audio workstations (DAWs) like Pro Tools or Logic Pro, which work natively with PCM WAV files
Frequently Asked Questions
No — the output WAV will be uncompressed, but it cannot recover quality that was lost when the audio was originally encoded to AAC inside the 3G2 file. AAC is a lossy codec, meaning some audio data was permanently discarded during mobile recording. The WAV will be a lossless container holding a lossless copy of the decoded AAC audio, so there's no further degradation, but the original compression artifacts remain. Think of it as freezing the quality at whatever level the 3G2 source had.
3G2 files use AAC audio at low bitrates (often 64–128 kbps) specifically designed for bandwidth-constrained CDMA mobile networks. PCM WAV at 16-bit/44.1kHz uses roughly 1,411 kbps — about 10 to 20 times more data for the same duration. There's no compression in pcm_s16le WAV; every audio sample is stored as a raw integer value. A 10-minute 3G2 clip with 128k AAC audio might produce a WAV file over 100MB.
Most 3G2 files use AAC (Advanced Audio Coding) as their default audio codec, which is what the 3GPP2 mobile standard specifies. Some older or variant 3G2 files may use AMR-NB or AMR-WB codecs, particularly for voice calls. If your 3G2 file uses AMR audio instead of AAC, FFmpeg will still decode it correctly and output it as pcm_s16le WAV — the command remains the same, but the resulting audio quality will reflect the very narrow bandwidth of AMR (optimized for speech, not music).
You can add the flags -ar for sample rate and -sample_fmt for bit depth. For example, to get 24-bit WAV at 48kHz, use: ffmpeg -i input.3g2 -vn -c:a pcm_s24le -ar 48000 output.wav. Common choices are pcm_s16le (16-bit, CD quality), pcm_s24le (24-bit, studio quality), and pcm_s32le (32-bit, mastering quality). Note that upsampling a 3G2 source won't add information that wasn't there, but matching your DAW's project settings can prevent unnecessary resampling downstream.
Yes — on Linux or macOS you can use a shell loop: for f in *.3g2; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.3g2}.wav"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This processes every 3G2 file in the current directory and outputs a matching WAV file. The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for batch workflows.
Partially. 3G2 files can carry metadata in their MP4-derived container (title, creation time, GPS coordinates from CDMA devices), but WAV has very limited metadata support compared to formats like FLAC or MP3. FFmpeg will attempt to copy basic tags into the WAV INFO chunk, but GPS coordinates, mobile-specific fields, and some 3GPP2 metadata atoms will be silently dropped. If metadata preservation is critical, consider extracting to FLAC instead, which supports rich Vorbis comment tags.
Technical Notes
3G2 is structurally derived from the MPEG-4 Part 12 container (like MP4 and MOV), so FFmpeg handles it reliably. The audio stream is almost universally AAC-LC at bitrates between 32k and 128k, reflecting the bandwidth constraints of early CDMA networks. The -vn flag is essential here because WAV is a pure audio container — attempting to write a video stream into WAV would cause FFmpeg to error out. The default output codec pcm_s16le produces standard CD-quality 16-bit little-endian PCM, which is universally readable by every audio application and operating system. WAV files have a 4GB file size limit due to the 32-bit chunk size field in the RIFF header; for very long 3G2 recordings, consider using the RF64 extension or switching the output to FLAC. No special flags like -movflags +faststart are needed for WAV since it has no streaming index structure. If the 3G2 file uses a non-standard audio codec (rare, but possible with some CDMA carrier variants), FFmpeg will still attempt decoding, though the fidelity of the result depends on codec support in the linked FFmpeg build.