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Supports: 3GP, 3G2
This page pulls the soundtrack out of a .3g2 clip — the 3GPP2 container CDMA phones recorded to — and writes it as FLAC, the Free Lossless Audio Codec. By the end you will know exactly which of the four controls change the result, why the FLAC comes out larger than the video it came from, and the one thing a lossless format genuinely cannot do for you: undo the phone codec that was applied years ago.
.3g2 onto the page or click "From my Computer" to browse. .3gp files from GSM-era handsets work on the same page, and several recordings can be queued together..flac. Files upload over an encrypted connection, are processed on our servers, and are deleted automatically after a few hours. No sign-up, no watermark.FLAC is lossless by definition, so none of these controls affects fidelity in the way a bitrate slider would on an MP3. They change size, speed and channel layout only.
| Control | Default | What it changes | When to move it |
|---|---|---|---|
| Compression level | 12 | Encoder effort, and therefore file size and processing time | Drop toward 1 if you are converting a long batch and want it done faster |
| Audio Channel | ORIGINAL | Mono or stereo layout of the output | Force MONO only if a downstream tool insists; a stereo copy of a mono source is twice the size for the same audio |
| Audio Sample Rate | ORIGINAL | Resamples the audio to 8000, 12000, 16000, 24000, 44100 or 48000 Hz | Only when a device or DAW demands a fixed rate. Upsampling never adds detail |
| Trim | Unchanged | Exports a start point and duration instead of the whole file | Whenever you want one section of a long recording |
Two of those deserve the arithmetic. Speech recorded by a CDMA handset is usually 8 kHz mono. Stored as 16-bit PCM that is 8,000 samples per second times 16 bits, or 128 kbit/s before FLAC's compression even begins — against an AMR-NB source that may have run as low as 4.75 kbit/s. That gap is why the FLAC is bigger than what it came from, and it is the format working correctly. Setting the sample rate to 44.1 kHz multiplies that raw figure by roughly five and a half without adding a single new frequency.
| What the 3G2 holds | What the FLAC gives you | |
|---|---|---|
| Coding | Lossy speech or music coding — AMR-NB, AMR-WB, AAC, or the CDMA voice codecs QCELP, EVRC and SMV | Lossless coding, defined by RFC 9639 (Standards Track, December 2024) |
| Typical sample rate | 8 kHz for narrowband speech codecs; 16 kHz for wideband variants | Whatever the source had, unless you change it |
| Frequency content | AMR-NB carries roughly 300-3400 Hz of speech band | Exactly the same band. Nothing above it is restored |
| Bitrate | As low as 4.75 kbit/s on AMR-NB | Typically an order of magnitude higher, because it is lossless |
| Further generation loss | Already one lossy generation deep | None from here on — every later FLAC-to-FLAC step is bit-exact |
So the honest summary is this: FLAC freezes the audio exactly as it exists today and guarantees no further degradation, which is the right thing to do with an irreplaceable recording. It does not and cannot rebuild what the handset's codec threw away in 2007.
.3g2.Two situations defeat any converter. Files copied off a failing memory card or interrupted mid-transfer are frequently truncated, and a 3GPP2 file with a damaged index will not decode past the break — trying it in VLC first tells you quickly whether the data is intact. And DRM-wrapped purchases from an old carrier deck are encrypted, not merely encoded; nothing on this site or elsewhere will convert those. If the audio you want is buried in a long recording, trimming a section before conversion is usually faster than exporting the whole thing and cutting it afterwards in the audio cutter.
No, and this is the point most worth understanding. FLAC is lossless, which means it reproduces its input bit-for-bit — but its input is already a lossy speech or music stream that a phone encoded years ago. FLAC preserves that perfectly and guarantees no further loss on any subsequent copy or conversion. What it cannot do is recreate frequencies and detail the original codec deleted. Think of it as archival packaging, not restoration.
Because it stops the bleeding. Every time you re-encode lossy audio into another lossy format, you lose a little more. Moving the recording into FLAC once means every future edit, copy and transcode starts from a bit-exact master rather than compounding the damage. That matters for recordings you cannot re-make: a voicemail from someone who has died, a field recording, a piece of family audio trapped on an old handset.
Simple arithmetic. Mono 8 kHz audio stored as 16-bit samples is 128 kbit/s before FLAC compresses it, while the AMR-NB stream it came from may have been running at 4.75 to 12.2 kbit/s. FLAC typically removes a meaningful share of that raw size, but it cannot approach a speech codec designed for a 2G-era radio link. A file several times larger than the source is the expected outcome, not a fault. To shrink it afterwards, run it through the FLAC compressor.
It controls how hard the encoder works to find redundancy, and nothing else. Level 12 is the top of the range and the default here; level 1 finishes faster and produces a larger file. The decoded audio is bit-identical at every level, so there is no quality decision to make — only a trade between processing time and a modest difference in file size. If you are converting a large batch and the wait is annoying, lower it.
No. Resampling 8 kHz speech to 44.1 kHz produces a file about five and a half times larger that contains exactly the same audible information, because the extra sample slots are interpolated from data that is already there rather than filled with newly captured sound. Leave "Audio Sample Rate" on ORIGINAL unless a specific piece of software refuses to load anything but 44.1 or 48 kHz.
3GPP2 files support the codecs 3GP does — AMR-NB, AMR-WB and AAC — plus the CDMA voice codecs that are the format's distinguishing feature: QCELP, EVRC, EVRC-B, EVRC-WB and SMV. Those last ones are narrowband speech codecs sampled at 8 kHz that very little software outside a handset can decode, which is a large part of why extracting the audio into a normal format is worth doing at all.
FLAC is the Free Lossless Audio Codec, developed under the Xiph.Org Foundation and published by the IETF as RFC 9639 on the Standards Track in December 2024. The format allows sample rates from 1 Hz to 1,048,575 Hz, bit depths from 4 to 32 bits and 1 to 8 channels, although most tools are tuned around CD parameters. It is handled by VLC, foobar2000, Audacity, most modern music software, and a wide range of current hardware players — but by no means everything, especially older car and hi-fi equipment.
Your 3G2 is uploaded over a TLS-encrypted connection, converted on our servers, and deleted automatically after a few hours. There is no sign-up, no watermark, and files are never shared or made public. If FLAC turns out not to suit the device you are targeting, the 3G2 converter hub lists the other audio and video formats this file can become.