Extract Audio from 3GP to WAV — Free Online Tool
Extract audio from 3GP mobile video files and save it as uncompressed WAV — converting the original AAC audio stream into lossless PCM (pcm_s16le) format. This is ideal when you need broadcast-ready, editing-friendly audio from footage recorded on older 3G-era phones.
to
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
3GP files typically contain AAC-encoded audio optimized for low-bandwidth mobile transmission. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed AAC audio, then re-encodes it as 16-bit signed little-endian PCM (pcm_s16le) — the standard uncompressed audio format used in WAV files. Because AAC is a lossy codec, the original compression artifacts are preserved in the decoded signal; no new quality is lost in the PCM encoding step, but the audio will not be 'better' than the original 3GP source. The resulting WAV file is significantly larger than the 3GP source because PCM stores every sample without compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, this runs as FFmpeg.wasm compiled to WebAssembly, executing entirely within your browser without sending your 3GP file to any server. |
-i input.3gp
|
Specifies the input 3GP file. FFmpeg reads the container structure, identifies the AAC audio stream and the video stream (typically H.264), and prepares to demux them for processing. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 or MJPEG video stream in the 3GP file. This is essential because WAV is a pure audio container and cannot hold video data. |
-c:a pcm_s16le
|
Decodes the AAC audio from the 3GP file and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio format used inside WAV files, compatible with virtually all audio editing tools and playback devices. |
output.wav
|
Defines the output file as a WAV container. FFmpeg writes the RIFF WAV header and streams the raw pcm_s16le audio data into it, producing an uncompressed audio file ready for editing, archiving, or broadcast use. |
Common Use Cases
- Recovering voice memos or phone call recordings captured on early-2000s or mid-range smartphones stored in 3GP format for use in audio editing software like Audacity or Adobe Audition
- Extracting spoken audio from 3GP video clips sent via MMS or downloaded from older mobile platforms to use as raw material for transcription services
- Pulling audio from 3GP footage recorded in the field for import into a DAW (Digital Audio Workstation) that requires uncompressed WAV input rather than AAC
- Archiving the audio track from a 3GP home video in an uncompressed format for long-term preservation before the container format becomes harder to play back natively
- Preparing audio from 3GP interview or lecture recordings for broadcast or podcast production workflows that require WAV at the ingest stage
- Isolating the audio from a 3GP music or performance clip to analyze waveforms or apply mastering processing without any additional lossy encoding step
Frequently Asked Questions
No — the WAV output will sound identical to the audio in the 3GP file, not better. The original audio was encoded as AAC at a low bitrate (commonly 64kbps or lower) to suit mobile bandwidth constraints, and that lossy compression is permanent. Converting to uncompressed PCM WAV faithfully represents the decoded AAC signal, but it cannot recover frequency information discarded during the original AAC encoding.
3GP uses AAC audio compression, which achieves small file sizes by discarding audio data the human ear is less sensitive to. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer — no compression at all. A one-minute AAC stream at 64kbps occupies roughly 480KB, while the equivalent uncompressed PCM WAV at 44.1kHz stereo occupies around 10MB. This size increase is expected and is the tradeoff for an editing-friendly, universally compatible audio file.
The output WAV inherits the sample rate of the AAC audio stream in the 3GP file. Mobile 3GP audio is often encoded at 8kHz or 16kHz (for voice) or 44.1kHz (for music), so the result depends entirely on your source file. The bit depth is fixed at 16 bits by the pcm_s16le codec used in this command. If you need a different bit depth — for example 24-bit for professional audio work — you can substitute pcm_s24le in the command.
3GP containers can carry limited metadata (title, author, creation date) using MP4-style atoms, but WAV has a very restricted metadata model based on the older RIFF INFO chunk standard. FFmpeg will attempt to map compatible tags, but many 3GP metadata fields have no WAV equivalent and will be silently dropped. If metadata preservation is critical, consider extracting to a format with richer tag support such as FLAC.
Replace pcm_s16le in the command with another PCM codec supported by WAV. For 24-bit audio use pcm_s24le, for 32-bit floating point use pcm_f32le, and for 8-bit use pcm_u8. The full substituted command would look like: ffmpeg -i input.3gp -vn -c:a pcm_s24le output.wav. Note that WAV does not support lossy codecs like AAC or MP3 in the way more modern containers do, so PCM or ADPCM variants are the practical choices.
Yes — on Linux or macOS you can run a shell loop: for f in *.3gp; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.3gp}.wav"; done. On Windows Command Prompt use: for %f in (*.3gp) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This applies the same -vn -c:a pcm_s16le logic to every 3GP file in the directory, outputting a matching WAV file for each. The browser-based tool on this page processes one file at a time.
Technical Notes
3GP was standardized by 3GPP (Release 6 and later) as a stripped-down derivative of the MPEG-4 Part 12 container, sharing its atom-based structure with MP4. Audio in 3GP files is almost universally AAC-LC (Low Complexity) because of its efficiency at very low bitrates — sometimes as low as 16kbps for voice-only recordings. When FFmpeg decodes this AAC stream and writes PCM WAV, it performs a full decode-encode cycle rather than a stream copy, because PCM and AAC are fundamentally incompatible at the bitstream level. The -vn flag ensures the H.264 or MJPEG video stream is completely ignored and not processed, which speeds up conversion and prevents errors on WAV output since WAV cannot contain video. The default pcm_s16le codec produces a standard 16-bit WAV file compatible with virtually all audio software, hardware samplers, and broadcast ingest systems. One known limitation is that 3GP files recorded on very old devices may have non-standard sample rates (such as 8000 Hz mono for voice calls), which WAV will faithfully reproduce — downstream software that expects 44.1kHz stereo may need to resample the result.