MP3 to WAV Converter

How to Convert MP3 to WAV Online

1. Upload Your MP3 Files: Drag and drop or click "+ Add Files" to select MP3 files from your device. Batch upload is supported — process dozens at once with the same settings.
2. Set Audio Sample Rate: Leave as Original to keep the source rate (typically 44.1 kHz for music), or pick 44.1 kHz for CD work, 48 kHz for video editing (Premiere, DaVinci Resolve, Final Cut), or 22.05 kHz / 16 kHz to shrink files for speech and voiceover use.
3. Set Audio Channel and Trim (Optional): Choose Mono to halve file size for podcasts and dictation, or keep Stereo for music. Use Trim to cut a clip — set start time and duration in HH:MM:SS.ms to extract a specific section without re-encoding the rest.
4. Convert and Download: Click Convert. Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically after one hour — no sign-up, no watermark, never shared. Download each WAV individually or grab a ZIP archive of the batch.

Why Convert MP3 to WAV?

MP3 is a lossy compression format — Fraunhofer's psychoacoustic encoder discards frequencies the ear is unlikely to perceive to hit 10:1 size ratios. WAV is a Microsoft/IBM RIFF container that almost always carries uncompressed linear PCM samples, the same raw format an audio CD uses. Converting MP3 to WAV unpacks the audio into the lossless container most editing and broadcast tools expect — it does not put back the data MP3 already threw away.

• Editing in a DAW — Ableton Live, Logic Pro, Pro Tools, FL Studio, Reaper, and Cubase all import WAV without re-decoding on every play, which keeps timing tight and avoids cumulative quality loss if you re-export the project. Each re-save of an MP3 project compresses again on top of an already-compressed source.
• Burning an audio CD — the Red Book CD-DA standard, defined by Sony and Philips in 1980, requires 44.1 kHz, 16-bit stereo PCM. Most CD burning software (iTunes, Windows Media Player, Nero) will accept MP3 input but internally decodes it to PCM first, so converting yourself gives you control over the result.
• Video and broadcast work — Premiere Pro, DaVinci Resolve, and Final Cut Pro handle WAV with frame-accurate sync; broadcast workflows standardize on 48 kHz / 24-bit per EBU and AES recommendations.
• Speech-to-text and ML pipelines — most ASR engines (Whisper, Google Speech-to-Text, AWS Transcribe) accept WAV directly and prefer 16 kHz mono PCM for voice models. Sending MP3 forces the service to decode first.
• Voice cloning, sampling, and mastering — plugins applied to lossy MP3 audio amplify existing compression artifacts (pre-echo, swirl on cymbals, smeared transients). Working in WAV gives the plugin clean samples even though the underlying recording is still bound by the MP3's information ceiling.
• Legacy hardware and embedded players — older car stereos, dictation transcribers, kiosk systems, and museum exhibits often only decode uncompressed PCM WAV.

MP3 vs WAV — Format Comparison

WAV File Size Cheat Sheet

WAV PCM file size is deterministic: bit depth × sample rate × channels ÷ 8 = bytes per second. Use this to predict storage before converting a long file.

Picking Sample Rate and Channels

Frequently Asked Questions

Does converting MP3 to WAV improve audio quality?

No. MP3 encoding permanently discards audio data the psychoacoustic model judges inaudible; that data is gone before you ever load the file into a converter. The output WAV will sound the same as the source MP3 — just larger and in an uncompressed container. The value of the conversion is workflow compatibility, not fidelity restoration.

Why is the WAV file roughly ten times larger than the MP3?

MP3 at 128 kbps targets about 1 MB per minute. CD-quality WAV runs at 1,411 kbps — roughly 10.6 MB per minute (1,411,200 bits/sec ÷ 8 ÷ 1,000,000 × 60). The factor of ~10 is exactly the compression ratio MP3 was designed to deliver. A 320 kbps MP3 produces a WAV about 4x larger; a 64 kbps MP3 produces one about 20x larger.

What sample rate and bit depth should I pick?

If you do not know, choose 44.1 kHz / 16-bit / Stereo — that is CD quality and the most universally compatible setting. Pick 48 kHz for anything destined for video. Going above the source MP3's sample rate (almost always 44.1 or 48 kHz) gives no quality benefit; it only wastes disk. See the cheat-sheet table above for exact file sizes.

Should I use the 16-bit (PCM_S16LE) or 24-bit (PCM_S24LE) codec option?

For converting MP3 to WAV, 16-bit is plenty — MP3's noise floor is well above 16-bit quantization noise, so 24-bit captures no useful extra information from a lossy source. Choose 24-bit only when the WAV is the start of a new mixing or mastering chain where headroom matters across many plugin stages.

Can I trim or cut a section of the MP3 during conversion?

Yes. Open Trim, set a start timestamp (HH:MM:SS.ms) and a duration, and the converter will export just that segment as WAV. For audio-only trimming without re-encoding effort spent on the rest of the file, the dedicated Audio Cutter is faster.

How do I convert WAV back to MP3 after editing?

After editing in a DAW, export the result and use WAV to MP3 to compress for distribution. Use 320 kbps CBR for archive copies, 192 kbps VBR for most listeners, or 128 kbps for podcasts and voice.

Is there a file size limit on the converter or the WAV format itself?

WAV itself is capped at 4 GiB because its RIFF header stores the file size in a 32-bit unsigned integer — that is about 6.8 hours of CD-quality stereo. For longer recordings, broadcast workflows use RF64 (EBU) or Sony's W64 instead, both of which use 64-bit size fields. XConvert processes files on its servers and deletes them automatically after one hour.

Will MP3 tags (artist, album, artwork) carry over to WAV?

Partially. WAV's standard "INFO" chunk supports basic fields (title, artist, year, comment) but not embedded artwork or extended ID3 tags. If full metadata matters for your library, FLAC is a better lossless target than WAV — it keeps tags and embedded artwork in roughly half the file size.

Why does Audacity warn about "importing uncompressed audio" being slow for MP3?

Audacity decompresses MP3 to PCM on import either by copying the decoded data into the project (fast, larger project folder) or by reading from the original file on every play (slow, smaller project). Converting to WAV first sidesteps that decision and removes the LAME decoder from the playback path entirely.