Convert 3G2 to AIFC — Free Online Tool
Convert 3G2 mobile video files to AIFC audio, extracting the AAC or MP3 audio track and re-encoding it as uncompressed PCM (pcm_s16be) in Apple's compressed AIFF container. This is ideal for recovering professional-quality audio from legacy CDMA mobile recordings for use in Apple-ecosystem audio workflows.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your 3G2 file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
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
3G2 files are multimedia containers designed for CDMA mobile networks, typically storing video encoded with H.264 (libx264) and audio encoded with AAC at low bitrates optimized for mobile transmission. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed AAC audio track, then re-encodes it as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFC container. AIFC is Apple's extension of AIFF that supports both uncompressed and compressed audio data. Because AAC is a lossy format, the original audio has already undergone quality reduction when the 3G2 was created — the PCM output preserves whatever fidelity remains in the AAC stream, but cannot recover information lost during the original mobile encoding.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, the same FFmpeg engine runs locally via WebAssembly (FFmpeg.wasm), so the command shown here is byte-for-byte identical to what you would run on a desktop installation. |
-i input.3g2
|
Specifies the input file — a 3G2 container holding H.264 video and compressed AAC audio encoded for CDMA mobile transmission. FFmpeg reads both streams but only the audio will be passed through to the AIFC output. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio format for AIFC and AIFF files. This decodes the lossy AAC audio from the 3G2 and re-encodes it as raw PCM samples, making the output immediately usable in Apple Logic Pro, GarageBand, and other AIFF-native tools without any additional decoding step. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For pcm_s16be this flag has no practical effect — the actual bitrate of uncompressed 16-bit PCM is fixed by sample rate and channel count — but it is included to maintain consistency with the tool's audio quality parameter system. |
output.aifc
|
Defines the output filename with the .aifc extension, which tells FFmpeg to use the AIFC container format. FFmpeg infers from this extension that the output should be wrapped in Apple's AIFF-C container, suitable for import into macOS and iOS audio applications. |
Common Use Cases
- Importing audio from old CDMA mobile phone video recordings into Apple Logic Pro or GarageBand, which natively support AIFC files
- Archiving audio from legacy 3G2 video clips captured on early smartphones or mobile cameras in a lossless PCM container for long-term preservation
- Extracting voice memos or spoken-word recordings stored in 3G2 format for editing in Final Cut Pro or other Apple-native post-production tools
- Converting mobile field recordings made on CDMA devices into AIFC for use as raw audio assets in professional broadcast or podcast production pipelines
- Recovering the audio track from 3G2 multimedia messages (MMS) or video attachments to use in audio restoration software that requires uncompressed input
- Preparing mobile-captured audio for mastering workflows where downstream tools expect big-endian PCM in an AIFF-family container
Frequently Asked Questions
No — the audio quality is fundamentally capped by the original AAC encoding inside the 3G2 file, which was compressed (lossily) when the mobile device recorded it. Converting to pcm_s16be AIFC produces an uncompressed PCM file, but it is a lossless copy of the already-lossy AAC data. You will get a larger file with no perceptible quality gain, but the AIFC format ensures no further degradation occurs in downstream processing.
3G2 files use AAC audio compression, which achieves very small file sizes by discarding audio data the human ear is less likely to notice. AIFC with pcm_s16be stores audio as raw, uncompressed 16-bit samples at the full sample rate — typically producing files 10 to 20 times larger for the same duration. For example, one minute of 128 kbps AAC audio in a 3G2 might be around 1 MB, while the equivalent pcm_s16be AIFC would be approximately 10 MB at 44.1 kHz stereo.
Yes. AIFC supports several PCM codecs including pcm_s24be (24-bit), pcm_s32be (32-bit integer), pcm_f32be (32-bit float), and pcm_f64be (64-bit float). To use 24-bit PCM, change the command to: ffmpeg -i input.3g2 -c:a pcm_s24be output.aifc. Keep in mind that because the source audio is AAC from a 3G2 file, higher bit depths will not recover lost audio data — they may be useful if the downstream application requires a specific bit depth.
3G2 files can carry metadata in their MP4-family container atoms, but AIFC's metadata support is limited compared to modern formats — it uses AIFF MARK and NAME chunks rather than ID3 or XMP tags. FFmpeg will attempt to map common metadata fields, but custom or mobile-specific tags from the 3G2 container may be dropped silently. If preserving metadata is important, verify the output with a tool like MediaInfo after conversion.
On macOS or Linux, you can loop through all 3G2 files in a directory with: for f in *.3g2; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.3g2}.aifc"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc". This is especially useful for batch-processing a library of old mobile recordings, and works identically to what the browser tool runs for each individual file.
For PCM codecs like pcm_s16be, the actual bitrate is determined entirely by the sample rate and channel count — not by the -b:a flag. A 44.1 kHz stereo pcm_s16be stream will always be approximately 1411 kbps regardless of what -b:a specifies. The flag is included in the command for structural consistency but is effectively ignored by FFmpeg when writing uncompressed PCM to AIFC.
Technical Notes
3G2 is a close relative of the MP4/ISOM container family, defined by the 3GPP2 standard for CDMA networks such as CDMA2000. Its audio streams are almost universally AAC-LC at low bitrates (often 32–128 kbps) because bandwidth efficiency was the primary design constraint. AIFC (Audio Interchange File Format Compressed) is Apple's extension of the original AIFF specification, adding support for compressed codecs alongside the classic uncompressed LPCM. When targeting pcm_s16be, the output is effectively uncompressed big-endian 16-bit audio — the same sample format used in standard Audio CDs and compatible with all Apple professional audio applications. One important limitation: if the 3G2 source was recorded in mono (common for voice calls or early mobile video), the AIFC output will also be mono; FFmpeg does not upmix channels by default. The 3G2 container does not support subtitles or chapters, so no non-audio stream data is at risk of being silently lost. Because AIFC uses big-endian byte ordering (unlike WAV's little-endian PCM), these files are not interchangeable with WAV despite both being uncompressed — some non-Apple tools may require conversion to AIFF or WAV for compatibility.