Convert 3GP to WAV — Free Online Tool
Convert 3GP mobile video files to WAV audio by extracting and decoding the AAC or AMR audio stream into uncompressed 16-bit PCM WAV format. Ideal for archiving mobile recordings or preparing audio for professional editing workflows that require lossless, broadcast-ready files.
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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
3GP files typically contain audio encoded as AAC (or sometimes AMR) at low bitrates optimized for 3G mobile networks. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track into raw, uncompressed PCM samples, then writes them into a WAV container using the pcm_s16le codec — 16-bit signed little-endian PCM. This means the audio is fully decoded from its lossy compressed state into an uncompressed representation. Because the source audio in 3GP is already lossy (AAC), the resulting WAV file will not recover any quality that was lost during the original 3GP encoding — it is simply an uncompressed copy of whatever audio data survived compression. No video stream, subtitle, or chapter data is carried over, since WAV is a pure audio container.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser via WebAssembly and on your local desktop command line. |
-i input.3gp
|
Specifies the input file — a 3GP container as used on 3G mobile phones, which typically holds a compressed video stream and a low-bitrate AAC or AMR audio stream. |
-c:a pcm_s16le
|
Sets the audio output codec to PCM signed 16-bit little-endian — the standard uncompressed audio encoding used in WAV files. This fully decodes the lossy AAC audio from the 3GP source into raw PCM samples with no further compression, making the file immediately usable in any audio editor or broadcast workflow. |
output.wav
|
Defines the output file as a WAV container. FFmpeg infers from the .wav extension that it should write a RIFF WAV file, and since WAV is a pure audio format, the video stream from the 3GP source is automatically dropped without needing an explicit '-vn' flag. |
Common Use Cases
- Extracting a voice memo or phone call recording saved as a 3GP file on an older Android or Nokia device so it can be imported into a DAW like Audacity or Adobe Audition for editing
- Preparing mobile field recordings captured on a 3G-era phone for use in broadcast or podcast production, where WAV is the required delivery format
- Archiving 3GP audio content in an uncompressed format to avoid further generation loss during future re-encoding steps
- Importing mobile video audio into video editing software (such as DaVinci Resolve or Final Cut Pro) that handles WAV tracks more natively than AAC-in-3GP
- Transcribing speech from a 3GP voice recording by feeding the extracted WAV file into speech-to-text tools that require uncompressed PCM input
- Stripping the audio from a 3GP MMS video message to use as a standalone sound clip or ringtone source in a compatible audio editor
Frequently Asked Questions
No — converting from 3GP to WAV does not recover any audio quality lost during the original 3GP encoding. The audio in a 3GP file is typically AAC-compressed at a low bitrate (often 32–64 kbps), which is inherently lossy. The WAV file produced is an uncompressed copy of that already-degraded audio. The benefit of WAV is not improved fidelity but rather that no additional quality is lost in any subsequent editing or re-encoding steps.
3GP files are heavily compressed, with audio stored at low bitrates using AAC — a codec specifically designed to minimize file size for mobile networks. WAV with pcm_s16le stores every audio sample as raw, uncompressed 16-bit integers with no compression whatsoever. A 1-minute 3GP audio track at 64 kbps would be around 480 KB compressed, but the equivalent 44.1 kHz stereo WAV would be roughly 10 MB. This size increase is expected and normal.
Yes — FFmpeg will use the sample rate present in the 3GP file's audio stream by default. 3GP files often contain audio sampled at 8 kHz (for voice calls) or up to 44.1 kHz for better-quality recordings. The output WAV will inherit whichever sample rate was used in the source, so if your 3GP voice recording was encoded at 8 kHz, the WAV will also be 8 kHz, which will sound narrow or telephone-like — this is a property of the source, not the conversion.
WAV supports several audio codecs including pcm_s24le (24-bit for higher dynamic range), pcm_s32le (32-bit float), pcm_u8 (8-bit unsigned, lowest quality), and even pcm_alaw or pcm_mulaw for telephony use. To change the codec, replace '-c:a pcm_s16le' in the FFmpeg command with your preferred option, for example: ffmpeg -i input.3gp -c:a pcm_s24le output.wav. For most purposes, pcm_s16le is the standard and most compatible choice.
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 is particularly useful when processing large batches of mobile recordings, since the browser-based tool handles one file at a time.
Yes, using FFmpeg's trim flags. Add '-ss' for a start time and '-t' for duration (or '-to' for an end time) before the output filename: ffmpeg -i input.3gp -c:a pcm_s16le -ss 00:00:10 -t 00:00:30 output.wav. This extracts 30 seconds of audio starting at the 10-second mark. This is useful when only a specific segment of a longer mobile recording is needed.
Technical Notes
3GP's audio stream is almost always AAC (MPEG-4 Audio) encoded at bitrates between 32 kbps and 128 kbps, sometimes lower for voice-optimized recordings using AMR-NB or AMR-WB codecs. FFmpeg handles AAC and AMR decoding natively, so no additional libraries are required for this conversion. The output WAV uses pcm_s16le — the most universally compatible PCM encoding, supported by every major DAW, operating system, and audio tool. One important limitation: WAV files do not support metadata fields like embedded album art or rich tagging (only basic RIFF INFO chunks), so any ID3-style tags present in the 3GP container will not be preserved in the output. Additionally, 3GP does not support multiple audio tracks, so there is no track selection ambiguity. If your 3GP file contains only an AMR audio stream rather than AAC, FFmpeg will still decode it correctly, but the resulting audio will reflect the very narrow 8 kHz bandwidth characteristic of AMR voice encoding.