Extract Audio from 3G2 to AIFC — Free Online Tool
Extract audio from 3G2 mobile video files and save it as AIFC, converting the AAC-encoded audio stream to uncompressed PCM (pcm_s16be) big-endian format. This is especially useful for recovering clean audio from CDMA-era mobile recordings in a professional, lossless container.
to
FFmpeg Command
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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 typically carry AAC audio alongside H.264 video, packaged for transmission over CDMA mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio, then re-encodes it as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFC container. Unlike a simple remux, this involves full audio transcoding: AAC is a lossy format, so the decoded PCM output represents the best possible quality recoverable from the compressed source — not a bit-perfect original. AIFC is Apple's compressed/uncompressed audio extension of AIFF, and the pcm_s16be codec produces standard uncompressed audio suitable for professional audio workflows, archiving, or import into DAWs.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (ffmpeg.wasm) — no data leaves your device. |
-i input.3g2
|
Specifies the input 3G2 file. FFmpeg reads the QuickTime-style atom structure of the 3G2 container to locate the AAC audio stream and H.264 video stream inside. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 video stream in the 3G2 file. Since AIFC is an audio-only format, this flag ensures only the audio track is processed. |
-c:a pcm_s16be
|
Decodes the AAC audio from the 3G2 source and re-encodes it as 16-bit signed big-endian PCM — the native uncompressed audio codec for the AIFC container format and the standard choice for Apple and professional audio compatibility. |
-b:a 128k
|
Sets a target audio bitrate of 128 kbps. For uncompressed PCM codecs like pcm_s16be, actual bitrate is determined by sample rate and bit depth rather than this parameter, so this flag has no practical effect on the output but is passed for consistency. |
output.aifc
|
Defines the output filename and signals FFmpeg to use the AIFC muxer based on the .aifc extension, wrapping the pcm_s16be audio stream in Apple's AIFF-C container format. |
Common Use Cases
- Recovering audio from old CDMA mobile phone recordings (e.g., flip phone videos) for archiving or transcription, where the 3G2 video is unwanted
- Importing audio from 3G2 footage into a Pro Tools or Logic Pro session via AIFC, which has native compatibility with Apple and professional audio environments
- Extracting voice memos or field recordings captured as 3G2 on legacy mobile devices to a lossless PCM format for forensic audio analysis
- Preparing audio from 3G2 conference or lecture recordings for further processing in audio editors that prefer uncompressed AIFF-family formats over compressed AAC
- Converting a library of archived 3G2 mobile videos to AIFC for long-term audio preservation, replacing the lossy AAC codec with uncompressed PCM in a standardized container
- Stripping audio from 3G2 sports or event clips recorded on early smartphones to use as sound effects in video production pipelines that require AIFF-compatible stems
Frequently Asked Questions
Not in the truest sense. The AIFC file will contain uncompressed PCM audio (pcm_s16be), which is itself a lossless format — but since the 3G2 source audio was already encoded with lossy AAC compression, some quality was lost at the recording stage. The conversion decodes the AAC as accurately as possible and stores that decoded audio without further degradation in the AIFC container. Think of it as lossless preservation of a lossy source, not a restoration of the original uncompressed signal.
AIFC does support compressed audio codecs in theory, but FFmpeg's AIFC muxer primarily supports PCM variants (pcm_s16be, pcm_s24be, pcm_alaw, pcm_mulaw, etc.) and does not support writing AAC into an AIFC container. Stream copying (-c:a copy) would fail here because AAC is not a valid codec for the AIFC format. The audio must be fully decoded from AAC and re-encoded as PCM, making transcoding unavoidable for this format pair.
The AIFC file will almost always be significantly larger than the 3G2 source, sometimes dramatically so. 3G2 files use AAC audio at low bitrates (often 64–128 kbps) combined with efficient H.264 video compression. The AIFC output stores only audio but as uncompressed 16-bit PCM, which for stereo audio at 44.1 kHz runs around 1.4 Mbps — roughly 10–20x the bitrate of a typical 3G2 AAC audio stream. A one-minute 3G2 clip might yield an AIFC file several times larger than the entire original video.
Yes, you can change the codec in the FFmpeg command to get higher bit-depth output. Replace '-c:a pcm_s16be' with '-c:a pcm_s24be' for 24-bit or '-c:a pcm_s32be' for 32-bit PCM — both are valid codecs for AIFC. However, since the source AAC in a 3G2 file was typically recorded at effective depths far below 24-bit, upscaling to a higher bit-depth won't recover lost detail; it will simply store the same decoded audio in a larger word size. 16-bit (pcm_s16be) is the practical sweet spot for this source material.
Metadata preservation between 3G2 and AIFC is limited. 3G2 stores metadata using the MPEG-4/QuickTime atom structure (title, date, GPS coordinates, etc.), while AIFC uses its own chunk-based metadata system with limited fields. FFmpeg will attempt to map compatible tags, but GPS data, device-specific fields, and extended 3GPP2 metadata are typically lost. If metadata preservation is critical, extract and document it separately before conversion using a tool like ExifTool.
To change bit-depth, swap the codec flag: use '-c:a pcm_s24be' for 24-bit, '-c:a pcm_alaw' for G.711 A-law telephony audio, or '-c:a pcm_mulaw' for mu-law — all are valid AIFC codecs. The '-b:a 128k' flag has minimal effect on uncompressed PCM codecs like pcm_s16be (bitrate is determined by sample rate and bit depth), so it can safely be omitted for PCM output. If you want to batch-process multiple 3G2 files on the command line, use a shell loop: 'for f in *.3g2; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.3g2}.aifc"; done'.
Technical Notes
3G2 is a variant of the MP4/MPEG-4 Part 12 container family, tailored for 3GPP2 and CDMA networks — it shares QuickTime-style atom structure with MPEG-4, which is why FFmpeg handles it similarly to MP4. The default audio codec in 3G2 is AAC (typically AAC-LC at low bitrates optimized for mobile transmission). AIFC (Audio Interchange File Format Compressed) is Apple's extension of AIFF that allows both uncompressed PCM and compressed codecs within the same container format, though in practice professional use favors the PCM variants. The pcm_s16be codec writes 16-bit signed PCM in big-endian byte order — the 'be' is significant because AIFF/AIFC is historically a big-endian format rooted in Motorola 68k and PowerPC architecture. The '-b:a 128k' flag in the command is effectively a no-op for pcm_s16be since uncompressed PCM bitrate is fixed by sample rate and channel count (not a quality parameter), but it does no harm. One known limitation: AIFC has no support for multi-channel surround audio beyond stereo in most implementations, though 3G2 files are virtually always mono or stereo anyway given their mobile origin. The output AIFC file will be compatible with macOS, iOS, and most professional DAWs including Logic Pro, Pro Tools, and Audacity.