Convert 3GPP to AIFF — Free Online Tool
Convert 3GPP mobile video files to AIFF, extracting the audio track and decoding it from AAC or MP3 into uncompressed 16-bit big-endian PCM — the lossless format used natively by macOS and professional audio applications. Ideal for recovering clean audio from mobile recordings when archival quality or DAW compatibility is the priority.
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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
3GPP files store audio using lossy codecs — typically AAC, sometimes MP3 — compressed for efficient transmission over 3G mobile networks at low bitrates (often as low as 32k–64k). During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio data, then re-encodes it into PCM signed 16-bit big-endian samples (pcm_s16be), which is the native PCM encoding used in AIFF files. This means the audio is fully decompressed into raw waveform data — no audio codec compression remains in the output. Because the source audio was originally lossy, the AIFF output is lossless in the sense that it contains no further compression artifacts, but it does preserve any quality loss that occurred when the original 3GPP file was created. The output file size will be dramatically larger than the source, as uncompressed PCM at 16-bit depth and standard sample rates requires far more storage than AAC-compressed audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles the demuxing of the 3GPP container, decoding of the compressed AAC or MP3 audio stream, and re-encoding into AIFF-compatible PCM. |
-i input.3gp
|
Specifies the input 3GPP file. FFmpeg reads the MP4-compatible container structure, identifies the audio stream (typically AAC), and ignores the video stream since the output format cannot hold video. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the standard uncompressed PCM format required by the AIFF specification. This fully decompresses the lossy AAC audio from the 3GPP source into raw waveform samples. |
output.aiff
|
Defines the output filename and triggers FFmpeg to use the AIFF container format, which wraps the pcm_s16be audio stream in Apple's Audio Interchange File Format structure, compatible with macOS, Logic Pro, GarageBand, and most professional DAWs. |
Common Use Cases
- Importing a voice memo or field recording captured on an older mobile phone into GarageBand or Logic Pro, which work natively with AIFF files on macOS
- Archiving audio from 3GPP video clips sent via MMS or downloaded from early-2000s mobile platforms, converting them to a lossless format before the source files degrade further
- Extracting interview audio recorded on a feature phone or early smartphone for use in a podcast editing workflow that requires uncompressed source files
- Preparing mobile-captured audio for forensic or legal analysis, where uncompressed AIFF is required by the review software or mandated by workflow standards
- Recovering audio from 3GPP video files shot on legacy Nokia, Sony Ericsson, or early Android devices to use as samples or loops in a DAW that requires lossless input
- Converting a series of 3GPP voiceover recordings to AIFF for import into an audio restoration tool like iZotope RX, which performs better analysis on uncompressed sources
Frequently Asked Questions
No — the audio in a 3GPP file was originally encoded with a lossy codec (typically AAC at 64k or lower), and that compression quality is permanently baked into the source. Converting to AIFF decompresses that audio into raw PCM waveform data, removing any further compression, but it cannot recover detail that was discarded during the original AAC or MP3 encoding. The AIFF file will be an exact lossless representation of the already-compressed audio — not better than the original, but with no additional degradation.
3GPP files use AAC audio compression, which can store a minute of audio in roughly 0.5–1 MB at typical mobile bitrates. Uncompressed PCM in AIFF stores every audio sample as a raw number — at 16-bit stereo and 44,100 Hz, that's about 10 MB per minute. The size increase is purely a result of removing compression, not adding any quality. If storage is a concern but you still need broad software compatibility, consider converting to FLAC instead, which is losslessly compressed.
No. AIFF is a pure audio format and cannot contain video streams. FFmpeg automatically drops the video track during this conversion and extracts only the audio. If you need to keep the video, you would need to target a video output format such as MP4 or MOV instead.
AIFF supports several PCM variants: pcm_s16be (16-bit, default), pcm_s24be (24-bit), pcm_s32be (32-bit), pcm_f32be (32-bit float), and pcm_f64be (64-bit float). To use a higher bit depth locally, modify the command to something like: ffmpeg -i input.3gp -c:a pcm_s24be output.aiff. For audio sourced from a low-bitrate 3GPP file, 16-bit is almost always sufficient since the source quality rarely justifies higher bit depths.
Yes. On macOS or Linux, you can run a simple shell loop: for f in *.3gp; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.3gp}.aiff"; done. On Windows Command Prompt, use: for %f in (*.3gp) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This processes each 3GPP file in the current directory and outputs a matching AIFF file, which is far faster than converting files one at a time.
3GPP files typically carry minimal metadata — usually just duration and occasionally a title tag stored in the MP4-style moov atom. AIFF supports a limited set of metadata via ID3 tags or AIFF chunks, but FFmpeg's conversion may not reliably map all 3GPP metadata fields across. If metadata preservation is critical, inspect the source tags first with ffprobe and add metadata manually using the -metadata flag in your FFmpeg command.
Technical Notes
The FFmpeg command for this conversion performs audio-only extraction and codec transcoding — there is no video remuxing involved since AIFF is an audio-only container. The source 3GPP container uses the same structural foundation as MP4 (ISO Base Media File Format), so FFmpeg can demux it reliably even for files created by legacy mobile hardware. The default output codec, pcm_s16be, uses big-endian byte ordering, which is the correct byte order for the AIFF specification (as opposed to AIFF-C or WAV, which use little-endian PCM). The sample rate of the output is inherited from the source audio stream — 3GPP audio is often encoded at 8,000 Hz (for voice calls), 16,000 Hz (wideband voice), or 44,100 Hz (music content), and FFmpeg preserves this rather than resampling. If the source was an 8 kHz mono voice track, the AIFF will reflect those characteristics. The tool does not apply any filters, normalization, or sample rate conversion, keeping the output a faithful PCM representation of whatever audio the 3GPP file contained. Because 3GPP does not support subtitles, chapters, or multiple audio tracks, there are no metadata structures to lose during conversion beyond basic container-level tags.