Convert MP3, M4A, OGG and FLAC to lossless WAV with your chosen bit depth and sample rate. Free — No Sign-Up
Decode any browser-supported format (MP3/M4A/OGG/FLAC…) and export a clean PCM WAV. Choose bit depth and (optional) sample rate.
Once your WAV is ready, make sure the track passes AI screening. Check it for AI-generated content and clean any artifacts before you send it to distributors or streaming platforms.
The artefactFX Audio Converter takes any file your browser can decode — MP3, M4A/AAC, OGG or FLAC — and writes it out as a clean, uncompressed PCM WAV. There's nothing to install: drop a file in, pick your settings, and download a lossless WAV built entirely on your device.
WAV matters because it's the universal, lossless format every DAW and mastering tool accepts. Bit depth (16 or 24-bit) controls headroom and noise floor, and sample rate (keep original, 44.1 or 48 kHz) controls timing resolution. Everything runs locally in your browser — your audio is never uploaded, so it's instant and completely private.
Three steps. A few seconds.
Load an MP3, M4A, OGG or FLAC. It's decoded to raw PCM right in your browser — nothing is uploaded.
Pick 16 or 24-bit and keep the original sample rate or resample to 44.1 or 48 kHz.
Save a lossless PCM WAV — built in the browser and downloaded straight to your device.
Simple settings, clean lossless output.
Load any format your browser can read and turn it into a standard WAV that every DAW and tool accepts.
Pick 16-bit CD quality for listening or 24-bit for editing, mixing and mastering headroom.
Keep the original sample rate, or resample to 44.1 or 48 kHz when a session or platform needs it.
The WAV is written from raw PCM in your browser — no re-compression, no watermark and nothing sent to a server.
Whenever you need a clean, compatible WAV.
Convert an MP3 sample or reference to WAV so your DAW imports it cleanly at the bit depth you need.
Hand your mastering engineer a 24-bit WAV at the right sample rate instead of a lossy file.
Keep a lossless WAV master of a track alongside its compressed copies for long-term storage.
When a tool won't accept your OGG, M4A or FLAC, convert it to plain WAV and it just works.
WAV is the closest thing audio has to a universal, no-compromise container. It stores raw, uncompressed PCM (pulse-code modulation) samples — the exact numbers your converter, DAW or mastering chain reads back, with nothing thrown away and nothing to decode. That makes it the format every digital audio workstation, mastering suite and distributor accepts without complaint. When you drop an MP3, M4A, OGG or FLAC into this converter, the audio is decoded to that raw PCM and written straight into a standard WAV, so whatever tool receives it sees clean, predictable samples instead of a compressed stream it has to unpack on the fly.
There are three everyday reasons producers reach for a WAV: editing (a DAW can scrub, slice and time-stretch uncompressed audio with far fewer surprises than a lossy file), mastering (engineers want the highest-fidelity source you can hand them), and archiving (a lossless master you can re-encode to anything later, forever). If you plan to do any real work on a track — not just listen to it — starting from WAV removes a whole category of problems before they happen.
This is the single most misunderstood point about audio conversion, so it is worth being blunt: converting an MP3 to WAV does not restore quality that the MP3 already discarded. Lossy formats like MP3, AAC/M4A and OGG Vorbis achieve their small file sizes by permanently deleting audio information a psychoacoustic model decides you are unlikely to hear — high-frequency detail, quiet sounds masked by louder ones, subtle stereo cues. Once that data is gone, it is gone. Wrapping the result in a WAV gives you a lossless copy of a lossy source: bit-for-bit faithful to the MP3, but no better than the MP3.
So why bother? Because a WAV of a lossy file is still genuinely useful. It stops generation loss — every time you re-encode MP3 to MP3 you shed a little more quality, whereas a WAV freezes the audio where it is. It gives editing tools a stable, uncompressed source to work on. And it produces a file every DAW and distributor will accept. FLAC is different: it is a lossless compressed format, so converting FLAC to WAV genuinely loses nothing at all — you are simply trading compression for universal compatibility.
Bit depth controls the dynamic range — the distance between the quietest detail and the loudest peak — and, with it, the noise floor. 16-bit gives roughly 96 dB of range, which is CD quality and comfortably beyond what most listening environments can resolve; it is the right choice for a finished file meant for playback or upload. 24-bit gives about 144 dB, far more headroom than any speaker or room actually needs — and that is precisely the point. The extra room lives at the bottom, so when you EQ, compress, add gain or stack effects in a DAW, rounding errors and quantization noise stay far below anything audible.
The practical rule: work in 24-bit, deliver final listening files in 16-bit. If you are importing into a session, prepping a file for a mixing or mastering engineer, or keeping a working master, choose 24-bit — this converter defaults to it for exactly that reason. If you just need a self-contained file to hand to someone or to play, 16-bit is smaller and indistinguishable in normal use. Note that converting a 16-bit source up to 24-bit will not add any real detail; it only prevents further loss during subsequent processing.
Sample rate is how many times per second the audio was measured, and it sets the highest frequency the file can represent (half the sample rate, by the Nyquist limit). The two rates you will meet almost everywhere are 44.1 kHz — the CD and music-streaming standard — and 48 kHz, the standard for video, film, broadcast and most game audio. Both capture the full range of human hearing with room to spare; the difference is mostly a matter of which world your project lives in, not audible quality.
Pick the target rate based on where the file is going. Delivering a single or an album for Spotify, Apple Music or a distributor? 44.1 kHz is the conventional choice. Cutting audio to picture, or handing stems to a video editor? Match the session at 48 kHz. If you are unsure and the file is not being placed against video, staying with the music-standard 44.1 kHz is the safe default. Want to inspect a track's tempo and key before committing to a session rate? Our free BPM & Key finder reads that straight from the audio.
Every time you change a file's sample rate, the software has to recalculate every single sample by interpolation. High-quality resampling is very good, but it is never perfectly transparent, and doing it repeatedly — or converting up and then back down — accumulates tiny artifacts for no benefit. That is why this converter defaults to "Keep original": unless something downstream demands a specific rate, the cleanest path is to leave the audio exactly as it was recorded or rendered.
Only resample when you have a concrete reason: your DAW session runs at a different rate and you want the import to sit at the project rate, a distributor or platform specification calls for 44.1 kHz, or a video pipeline requires 48 kHz. Resampling once, deliberately, at the point of delivery is fine. Resampling casually, back and forth, is a habit worth breaking.
Plenty of workflows will simply refuse anything but an uncompressed file. A few of the common ones:
If you are prepping a release, converting to WAV is usually one step in a short chain. It pairs naturally with trimming a file down with our audio cutter, checking your levels on the LUFS meter, and — for AI-assisted music — screening the track with the AI Checker and, if needed, tidying it with the AI Cleaner before it goes out.
Lossless quality is not free — it is large. Because WAV stores every sample uncompressed, size scales directly with sample rate, bit depth, channel count and length. A stereo 44.1 kHz / 16-bit WAV runs about 10 MB per minute; bump it to 24-bit and it grows by half; move to 48 kHz and it grows again. A three-minute stereo track lands somewhere around 30–50 MB depending on your settings, versus a few megabytes for the same track as an MP3.
That is the whole point of the format, and the reason to be deliberate: use WAV where fidelity matters — masters, session imports, deliverables — and keep lightweight lossy copies for casual sharing and previews. Choosing 16-bit and 44.1 kHz for a listening file, or 24-bit and your session rate for working material, lets you spend those megabytes only where they buy you something.
This converter runs entirely on your own device. The moment you drop a file in, the browser decodes it to PCM locally and builds the WAV in memory — nothing is uploaded to a server. That means it is fast (no waiting on an upload or a download of the result), it works on material you would never want to send to a stranger's cloud, and there is no queue, no watermark, no account and no cap on how many files you run. Your audio never leaves your computer, which is the simplest possible privacy guarantee. One honest caveat: this tool exports WAV only today — MP3 and FLAC encoding need an extra encoder library and are planned, but WAV export works right now.
Everything you need to know about converting audio to WAV online.