How to convert wav to flac — lossless, ~50% smaller
- Step 1Drop your WAV file onto the tool — This converter takes a single file (
acceptsMultipleis false). It reads the input extension from the file itself, so a correctly-tagged.wavis decoded by FFmpeg's WAV demuxer. The file stays in the browser tab — it is written to FFmpeg's in-memory WASM filesystem, not sent anywhere. - Step 2Pick a compression level — The Compression level dropdown offers
0 · fastest,5 · balanced,8 · default sweet spot, and12 · smallest, slowest. The default is 8. Leave it on 8 unless you have a specific reason — it is the standard archival setting and the difference in file size between 8 and 12 is usually 1-3%. - Step 3Understand that the level never touches audio quality — Levels 0 through 12 all produce mathematically identical decoded PCM. A higher level just asks the encoder to search harder for efficient prediction coefficients, so it takes longer and shaves a little more size. There is no perceptual setting here — FLAC is lossless at every level.
- Step 4Run the conversion — FFmpeg transcodes with the native
flacencoder and-compression_level <n>. No-b:abitrate is passed — bitrate is a lossy concept and is meaningless for FLAC. Encode speed scales with the level and your CPU; a few-minute stereo track is near-instant, a long 24-bit master at level 12 takes longer. - Step 5Download the .flac — The output is a
.flacfile with MIMEaudio/flac, same channel count, sample rate, and bit depth as the WAV. The filename keeps your base name with a.flacextension. - Step 6Verify losslessness if it matters to you — Decode the FLAC back to WAV in any player and compare — or trust the embedded MD5.
flac --test file.flac(the reference CLI) re-decodes and checks that MD5 against the stream, confirming a clean round-trip without needing the original WAV.
Compression level: what actually changes
The four levels exposed in the dropdown, what each does, and what it does NOT do. Sizes are typical ranges versus the source WAV; exact savings depend entirely on the audio.
| Level (dropdown label) | Encoder effort | Typical size vs. WAV | Decoded audio |
|---|---|---|---|
| 0 · fastest | Minimal prediction search; quickest encode | ~50-65% of WAV | Bit-identical |
| 5 · balanced | Moderate search (FLAC reference default) | ~45-60% of WAV | Bit-identical |
| 8 · default sweet spot | High effort; this tool's default | ~42-58% of WAV | Bit-identical |
| 12 · smallest, slowest | Exhaustive apodization search (FFmpeg-specific, beyond the reference 0-8) | ~41-57% of WAV | Bit-identical |
WAV source → FLAC: what is and is not preserved
The transcode decodes PCM and re-encodes losslessly. Sample data and most metadata survive; container-only WAV oddities do not carry into FLAC.
| Property | Behaviour through the converter |
|---|---|
| PCM sample values | Preserved exactly — lossless round-trip, MD5-verifiable |
| Bit depth (8/16/24-bit) | Preserved; FLAC stores the native depth (32-bit float WAV is converted to integer PCM — see edge cases) |
| Sample rate / channel count | Preserved unchanged (no resampling, no down-mix) |
| Tags / metadata | Carried via -map_metadata 0 where the WAV holds INFO/BWF tags |
| Embedded cover art | Re-attached as a FLAC picture block if the source WAV contains one |
| WAV-specific chunks (cue, BWF bext timecode) | Not guaranteed — standard FLAC has no slot for proprietary WAV chunks; treat the WAV as the chunk-level master |
Cookbook
Concrete size and speed outcomes from real WAV sources. Sizes are illustrative of typical compression behaviour, not guarantees — your mileage depends on the audio content.
Standard CD-quality track at the default level 8
The everyday case: a 16-bit/44.1 kHz stereo song. Level 8 is the default and the right call for almost everyone.
Source: song.wav 16-bit / 44.1 kHz / stereo / 4:10 ~42 MB Level: 8 (default) Output: song.flac ~25 MB Result: ~40% smaller, bit-identical on decode
Level 0 vs 12 on the same file — speed/size trade, identical audio
Demonstrates the core point: the level moves size and encode time, never quality. The decoded PCM is byte-for-byte the same.
Source: master.wav 24-bit / 96 kHz / stereo / 5:00 Level 0: encodes fast -> ~96 MB Level 12: encodes slower -> ~91 MB Decode both back to WAV: md5(level0.wav) == md5(level12.wav) == md5(master.wav) Same audio. You only chose how hard the encoder worked.
Mono voice recording compresses harder than a dense mix
Sparse, low-entropy audio (single voice, silence, acoustic) compresses far better than a loud, full-spectrum master. Same level, very different savings.
Mono interview.wav 16-bit / 48 kHz / mono / 30:00 ~165 MB -> level 8 -> ~70 MB (~58% smaller) Mastered EDM track.wav 16-bit/44.1k/stereo/3:30 ~37 MB -> level 8 -> ~28 MB (~24% smaller)
Hi-res 24-bit master kept lossless for the library
FLAC stores 24-bit natively, so a hi-res master shrinks without any conversion to 16-bit. Bit depth is preserved.
Source: album_master.wav 24-bit / 88.2 kHz / stereo Level: 8 Output: album_master.flac 24-bit / 88.2 kHz / stereo No down-conversion. ffprobe shows: sample_fmt=s32 (24 in 32), same sample_rate, lossless.
Round-trip proof for a paranoid archive ingest
If you must prove the conversion lost nothing, decode the FLAC and compare hashes against the original WAV. The reference CLI does this with one command.
# After downloading song.flac from this tool: flac --test song.flac song.flac: ok <- internal MD5 matches decoded PCM # Or compare to the original yourself: flac -d song.flac -o roundtrip.wav fc /b song.wav roundtrip.wav (Windows) -> no differences
Edge cases and what actually happens
32-bit float WAV source
Converted to integerSome DAWs export 32-bit floating-point WAV. FLAC's standard format stores integer PCM, so a 32-bit float source is converted to integer samples by the encoder. The result is still lossless relative to the integer representation, but it is not a bit-identical container for float data. If you specifically need to preserve 32-bit float exactly, keep the WAV master alongside the FLAC.
Free tier file over 50 MB
413 blockedThe audio family Free tier caps a single file at 50 MB. A long or hi-res WAV easily exceeds that — a 10-minute 24-bit/96 kHz stereo file is ~330 MB. Pro raises the cap to 200 MB, and Pro-media / Developer to 100 GB. The limit is on the input WAV size, not the FLAC output.
WAV longer than the Free tier duration cap
RejectedAudio also has a per-file duration limit, separate from size: Free is 30 minutes, Pro is 120 minutes, Pro-media and Developer are unlimited. A 45-minute lecture WAV that happens to be under 50 MB (e.g. mono, low rate) is still blocked on Free because it exceeds 30 minutes.
Uploading multiple files at once
Single file onlyThis converter has acceptsMultiple set to false — it processes one WAV per run. For a folder of WAVs, convert them one at a time, or use a Pro plan workflow. There is no in-browser drag-reorder or queue in this tool.
FLAC came out the same size or barely smaller
ExpectedFLAC's savings depend on the audio's redundancy. A heavily-limited, full-spectrum master (modern pop/EDM at maximum loudness) has high entropy and may only shrink 15-25%. That is normal and still lossless — there is no setting that compresses incompressible noise further without losing data.
Output is larger than expected for a quiet file
By designFLAC has fixed per-frame overhead. For very short clips or pathological inputs the overhead can dominate, but for real-world tracks the savings always outweigh it. Raising the compression level helps only marginally; it is not a fix for tiny files.
BWF (Broadcast WAV) timecode / bext chunk
Not carriedBroadcast WAV stores production timecode and origination metadata in a bext chunk. Standard FLAC has no equivalent slot, so that chunk is not preserved. For film/broadcast post where bext timecode is load-bearing, keep the BWF master; use FLAC only for size-reduced reference copies.
Input file is not actually a WAV
Decoded if FFmpeg can read itFFmpeg reads the real format from the file, not the name. If you feed an AIFF or other PCM file mislabelled as .wav, FFmpeg may still decode and FLAC-encode it losslessly. A genuinely corrupt or truncated file fails during decode with an FFmpeg error rather than producing a silent bad output.
Already-lossy source 'converted' to FLAC
No quality gainIf your WAV was itself decoded from an MP3, the lossy artifacts are baked into the PCM. FLAC will store them losslessly but cannot recover what was discarded. FLAC freezes current quality; it never restores it.
Browser tab closed mid-encode on a huge master
Run lostBecause everything runs in the tab, closing or reloading it during a long level-12 encode of a multi-hundred-MB master discards the in-progress job. Nothing was uploaded, so nothing is left server-side either — just re-run it and keep the tab open.
Frequently asked questions
Is FLAC really lossless compared to the WAV?
Yes. FLAC is mathematically reversible — the encoder applies entropy coding to the PCM and the decoder reconstructs the exact same samples. FFmpeg's flac encoder also stores an MD5 of the decoded audio in the header, so flac --test can prove the round-trip is bit-identical without the original WAV present.
What compression level should I pick?
Leave it on 8, the default. The dropdown offers 0 (fastest), 5 (balanced), 8 (default sweet spot), and 12 (smallest, slowest). Level affects only encode time and file size — every level decodes to identical audio. Going from 8 to 12 typically saves another 1-3% for noticeably longer encode time.
Does a higher compression level reduce audio quality?
No — and this is the single most common misconception about FLAC. The level controls how hard the encoder searches for efficient prediction, not what audio it keeps. Levels 0 and 12 produce byte-for-byte identical decoded PCM. There is no perceptual quality knob on a lossless codec.
How much smaller will my file get?
Typically 30-60% smaller, but it depends entirely on the audio. Sparse content (single voice, acoustic, lots of silence) compresses hardest; dense, loud, full-spectrum masters compress least. FLAC cannot compress high-entropy material as much, because there is less redundancy to exploit.
Is the maximum level really 12, not 8?
The FLAC reference encoder documents 0-8. FFmpeg's flac encoder — which this tool uses — extends the range to 12 with additional apodization search. The engine accepts 0-12 and the dropdown exposes 0, 5, 8, and 12. Levels above 8 are FFmpeg-specific and still fully lossless and standard-decodable.
Does this preserve my 24-bit or hi-res master?
Yes. FLAC natively stores 8, 16, and 24-bit PCM at any sample rate. A 24-bit/96 kHz WAV becomes a 24-bit/96 kHz FLAC — no down-conversion to 16-bit, no resampling. The one exception is 32-bit float WAV, which is converted to integer PCM (see the edge cases).
Are my files uploaded anywhere?
No. The entire conversion runs in your browser tab via FFmpeg 8.1 compiled to WebAssembly. Your WAV is written to an in-memory filesystem, encoded, and handed back as a download. Nothing is transmitted to a server, which also means there is no upload wait on large masters.
Why is there no bitrate option?
Bitrate is a lossy concept — it describes how much data a codec is allowed to throw away. FLAC throws nothing away, so it has no target bitrate. The tool deliberately passes no -b:a for FLAC; the only relevant control is compression level, which trades encode effort for size.
Will my metadata and album art survive?
Tags carry across via -map_metadata 0 where the WAV holds them, and embedded cover art is re-attached as a FLAC picture block. Note that plain WAV often carries little metadata to begin with; proprietary WAV chunks like BWF bext timecode have no FLAC equivalent and are not preserved.
Can I convert FLAC back to WAV or to MP3 later?
FLAC decodes back to bit-identical WAV in any player or with flac -d. For a smaller lossy share copy, use the sibling FLAC to MP3 tool. To go the other lossless direction from a compressed source, see MP3 to FLAC (archival container only — it cannot restore lossy data).
How big a WAV can I convert?
Free tier allows up to 50 MB and 30 minutes per file; Pro allows 200 MB and 120 minutes; Pro-media and Developer allow up to 100 GB with unlimited duration. Both the size and the duration limit apply — a long file under the size cap can still be blocked by the duration cap.
What's the difference between this and the WAV to MP3 tool?
WAV to FLAC is lossless — same audio, ~half the size, best for masters and archives. WAV to MP3 is lossy — much smaller, perfect for distribution and hosting, but it discards data. Choose FLAC when you need to keep the exact audio; choose MP3 when small size matters more than perfect fidelity.
Privacy first
Every JAD Audio tool runs entirely in your browser via FFmpeg (WebAssembly) and RNNoise. Your audio files never leave your device — verified by zero outbound network requests during processing.