Convert 3GPP to AIFC — Free Online Tool
Convert 3GPP mobile video files to AIFC audio, extracting the AAC or MP3 audio track and re-encoding it as 16-bit big-endian PCM for professional audio workflows. AIFC's uncompressed PCM output gives you a lossless-quality working copy suitable for audio editing software like Pro Tools, Logic Pro, and Audacity.
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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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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
3GPP files store audio in compressed lossy formats — typically AAC (the default) or MP3 — inside a mobile-optimized container. During this conversion, FFmpeg strips the video track entirely and decodes the compressed audio stream from the 3GP container, then re-encodes it as PCM signed 16-bit big-endian audio (pcm_s16be) wrapped in an AIFC container. Because the source audio is already lossy (AAC or MP3), this is a lossy-to-PCM transcode: the PCM output will be large and uncompressed, but it cannot recover detail lost in the original mobile encoding. The result is a high-fidelity working copy of whatever audio quality existed in the 3GPP file, now stored in a format that professional audio tools can import and edit without further degradation.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In the browser-based tool this runs as FFmpeg.wasm compiled to WebAssembly, executing entirely client-side without any server upload. |
-i input.3gp
|
Specifies the input file in 3GPP format. FFmpeg reads the container, identifies the audio stream (typically AAC or MP3) and any video stream, and makes both available for mapping into the output. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, the standard uncompressed audio encoding for AIFC files. FFmpeg decodes the lossy AAC (or MP3) audio from the 3GPP source and re-encodes each sample as a raw 16-bit big-endian integer, discarding the lossy compression. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For the pcm_s16be codec this flag is effectively ignored — uncompressed PCM bitrate is fixed by sample rate and channel count — but it is included in the command template and does not cause any errors. |
output.aifc
|
Defines the output filename with the .aifc extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an AIFC container. The video track from the 3GPP input is automatically dropped because AIFC is a pure audio format. |
Common Use Cases
- Importing a voicemail or voice memo recorded on an older 3G mobile phone into Pro Tools or Logic Pro, which require AIFC/AIFF format for native project import
- Extracting the audio from a 3GPP video clip captured on a feature phone to use as a raw audio sample in a music production workflow
- Archiving mobile audio recordings from legacy 3G devices in an uncompressed PCM format that will remain readable by audio software decades from now
- Preparing 3GPP interview recordings for professional podcast editing, converting to AIFC so editors can work without lossy re-encoding in their DAW session
- Converting 3GPP audio to AIFC as an intermediate step before applying mastering or noise reduction, since uncompressed PCM preserves headroom for processing plugins
- Submitting audio recorded on a mobile device to a broadcast or film post-production pipeline that mandates AIFC delivery format
Frequently Asked Questions
No — the conversion will not improve quality beyond what was captured in the original 3GPP file. The audio in 3GPP is stored in a lossy format (usually AAC at a low bitrate like 64k, optimized for mobile networks), and that compression is irreversible. The AIFC output will be uncompressed PCM, meaning no further quality is lost during editing, but the ceiling is set by the original mobile recording. Think of it as a very large, editable copy of a compressed original.
3GPP files use highly efficient lossy compression (AAC or MP3) designed for bandwidth-constrained 3G mobile networks, resulting in very small file sizes. AIFC with pcm_s16be stores raw uncompressed audio samples — at 16-bit stereo 44.1kHz that is roughly 10MB per minute, compared to under 1MB per minute for AAC at 64k. The size difference reflects the absence of compression, not a gain in quality.
The video track is discarded entirely. FFmpeg reads the 3GPP container, selects only the audio stream, and writes an AIFC file, which is a pure audio format with no video container support. If you need to keep the video, you should convert to a video output format instead. No video data is written to or referenced by the AIFC output.
AIFC descends from Apple's AIFF format, which was designed on Motorola 68000-based Macintosh hardware that used big-endian byte order. The pcm_s16be codec reflects this heritage and remains the standard for AIFC files expected by professional Apple audio software and many broadcast systems. WAV by contrast uses little-endian PCM because it originated on Intel x86 platforms. Both represent the same underlying audio data — the byte order only matters for software reading the raw bytes.
You can replace '-c:a pcm_s16be' with another codec supported by AIFC, such as 'pcm_s24be' for 24-bit depth, 'pcm_s32be' for 32-bit integer, 'pcm_f32be' for 32-bit float, or 'pcm_alaw' / 'pcm_mulaw' for telephony-grade compressed audio. For example: 'ffmpeg -i input.3gp -c:a pcm_s24be output.aifc' produces a 24-bit AIFC file. Note that since the 3GPP source is typically 16-bit AAC, encoding to pcm_s24be does not add real bit depth — it simply stores the same data in a wider container.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.3gp; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.3gp}.aifc"; done'. On Windows Command Prompt use: 'for %f in (*.3gp) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc"'. Each file is processed sequentially and output with the same base filename and an .aifc extension. This is especially useful for converting a folder of archived mobile recordings in one step.
Technical Notes
The 3GPP container format was designed for transmission over 3G mobile networks, so its audio tracks are almost universally encoded at low bitrates — AAC at 64k is the typical default, though some devices used AMR-NB for voice-only recordings. FFmpeg can decode all common 3GPP audio codecs into PCM for the AIFC output, but AMR-NB decoding requires FFmpeg to be built with non-free codec support, which may not be available in all browser-based builds. AIFC supports both lossy (alaw, mulaw) and lossless (PCM) codecs in the same container format — this tool defaults to pcm_s16be, which is lossless at 16-bit depth and universally supported by professional audio software on macOS and beyond. Metadata embedded in the 3GPP container (such as creation date or GPS tags from some phones) is not carried into AIFC, as the format has no equivalent metadata fields. The '-b:a 128k' flag in the command has no practical effect on uncompressed PCM codecs since PCM bitrate is determined entirely by sample rate and bit depth, but it is included for consistency and ignored by FFmpeg when writing pcm_s16be. If your 3GPP file contains only video and no audio track (uncommon but possible), FFmpeg will produce an empty or error output — verify the source file has an audio stream first using 'ffmpeg -i input.3gp'.