Convert 3GPP to WAV — Free Online Tool
Convert 3GPP mobile video files to WAV audio by extracting and decoding the AAC or MP3 audio track into uncompressed PCM audio. Ideal for pulling high-fidelity audio from mobile recordings when you need broadcast-ready or lossless WAV files for editing, archiving, or professional use.
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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 typically contain video encoded with H.264 and audio encoded with AAC (or occasionally MP3) — both lossy compressed formats designed to minimize file size on mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed AAC audio track into raw, uncompressed PCM audio encoded as signed 16-bit little-endian samples (pcm_s16le), which is the standard WAV format. Because the audio must be fully decoded from AAC and re-encoded into PCM, this is a transcoding operation — not a lossless remux. Any quality loss that occurred when the 3GPP file was originally recorded is preserved, but no additional lossy compression is introduced in the output WAV file. The resulting WAV file will be significantly larger than the source 3GPP because PCM audio carries no compression whatsoever.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles the decoding of the 3GPP container and its AAC audio track and the encoding of the output PCM WAV file. |
-i input.3gp
|
Specifies the input file — a 3GPP container typically holding an H.264 video stream and an AAC audio stream recorded by a mobile device. FFmpeg will demux this container to access its individual streams. |
-c:a pcm_s16le
|
Instructs FFmpeg to decode the AAC audio from the 3GPP file and re-encode it as signed 16-bit little-endian PCM — the standard uncompressed audio codec used in WAV files and the most widely compatible option for downstream audio software and hardware. |
output.wav
|
Defines the output file as a WAV container. FFmpeg infers from the .wav extension that the output should be wrapped in the Waveform Audio File Format, which pairs correctly with the pcm_s16le codec specified in the command. |
Common Use Cases
- Extracting a voice memo or interview recorded on a mobile phone in 3GPP format for import into a professional audio editor like Audacity, Adobe Audition, or Pro Tools
- Converting a 3G mobile video of a live musical performance to WAV so the audio can be used as a stem in a music production project without further lossy compression
- Preparing mobile-recorded field audio for broadcast submission, where WAV PCM is required as the delivery format by radio or TV stations
- Archiving audio from old 3GPP mobile recordings in an uncompressed format to ensure long-term accessibility without codec dependency
- Stripping the audio from a 3GPP video captured on a legacy handset to transcribe speech, where WAV is required by transcription software APIs
- Recovering audio from 3GPP video clips to use as sound effects or ambient recordings in video post-production workflows
Frequently Asked Questions
No — converting 3GPP to WAV does not recover any quality lost during the original AAC encoding. The AAC audio in the 3GPP file is already lossy-compressed, and that compression is irreversible. What converting to WAV does provide is an uncompressed container that introduces no additional quality loss, making it a stable and neutral format for editing or archiving the audio exactly as it existed in the 3GPP file.
3GPP stores audio using AAC compression, which achieves very high compression ratios — typically encoding audio at 64kbps or less for mobile use. WAV with PCM audio is completely uncompressed, storing every audio sample as raw binary data. A 1-minute AAC audio track at 64kbps might occupy about 480KB in a 3GPP file, but the same audio decoded to 16-bit PCM at 44.1kHz stereo in WAV will be approximately 10MB. This size difference is expected and normal.
Metadata preservation from 3GPP to WAV is limited. 3GPP files can carry metadata in their MP4-style container atoms, but WAV has a very minimal metadata structure (INFO chunks or ID3 tags, depending on the application). FFmpeg will attempt to map compatible metadata fields, but container-specific tags like GPS coordinates, device model, or creation timestamps from the mobile camera are likely to be dropped. If metadata preservation is critical, inspect the output file with a tool like MediaInfo after conversion.
Most 3GPP files recorded by mobile phones use AAC (Advanced Audio Coding), typically at low bitrates between 32kbps and 128kbps. Some older devices used AMR (Adaptive Multi-Rate), though AMR is less common in FFmpeg-compatible 3GPP files. The original AAC bitrate directly determines the ceiling of audio quality in the output WAV — a 32kbps AAC source will produce a WAV that sounds like 32kbps AAC, because the PCM format simply holds the decoded result of that lossy source without any form of quality enhancement.
To produce a 24-bit WAV instead of the default 16-bit, replace pcm_s16le with pcm_s24le in the command: ffmpeg -i input.3gp -c:a pcm_s24le output.wav. You can also use pcm_s32le for 32-bit integer or pcm_f32le for 32-bit float PCM. Note that since the source audio is lossy AAC, increasing bit depth will not recover lost audio information — it simply changes the numerical precision of the decoded samples in the output file, which may be required by certain professional audio workflows or DAWs.
Yes. On Linux or macOS, you can use a shell loop: for f in *.3gp; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.3gp}.wav"; done. On Windows Command Prompt, use: for %f in (*.3gp) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This processes each 3GPP file in the current directory and outputs a corresponding WAV file with the same base filename. The browser-based tool on this page processes one file at a time, so the FFmpeg command is especially useful for batch workflows involving many files.
Technical Notes
The default output codec pcm_s16le produces WAV files with signed 16-bit little-endian PCM samples, which is the most universally compatible WAV variant — supported by virtually all audio software, operating systems, and hardware devices. The sample rate of the output WAV is inherited from the source 3GPP audio track; FFmpeg does not resample unless explicitly instructed to. Mobile 3GPP recordings are commonly captured at 8kHz (voice calls), 16kHz (wideband voice), or 44.1kHz/48kHz (music or video). If downstream software requires a specific sample rate, add -ar 44100 or -ar 48000 to the command. Since WAV does not support multiple audio tracks, only the first audio stream from the 3GPP file will be extracted. 3GPP files with AMR audio (common in very old handsets) may require additional FFmpeg build flags; the AAC path handled by this tool is the standard modern case. The -movflags +faststart flag present in 3GPP input handling is relevant only for streaming optimization of the source container and has no effect on the WAV output, which carries no container-level streaming metadata.