Convert MP3 to WAV — Free Online Tool
Convert MP3 files to WAV format by decoding the lossy MPEG Audio Layer III compression into uncompressed PCM audio (pcm_s16le, 16-bit signed little-endian). The resulting WAV file is significantly larger but fully uncompressed, making it ideal for audio editing, broadcast workflows, and any application requiring raw PCM audio data.
to
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
MP3 files store audio using perceptual lossy compression via the MPEG Audio Layer III codec, which permanently discards audio information deemed inaudible to human ears. During this conversion, FFmpeg's libmp3lame decoder fully decodes the MP3 bitstream back into raw PCM audio samples, then writes those samples into a WAV container using the pcm_s16le codec — 16-bit signed integers in little-endian byte order, which is the standard uncompressed WAV format. It is important to understand that this process does not recover the audio data lost during the original MP3 encoding: the WAV output will be uncompressed, but its true audio fidelity is bounded by the quality of the source MP3. No re-encoding or further quality degradation occurs — the decoded PCM data is written directly to the WAV file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, the underlying engine that powers this browser-based conversion via its WebAssembly (FFmpeg.wasm) port. |
-i input.mp3
|
Specifies the input file as an MP3, telling FFmpeg to use its libmp3lame-compatible MP3 decoder to read and decode the MPEG Audio Layer III bitstream into raw PCM audio samples for processing. |
-c:a pcm_s16le
|
Sets the audio codec for the output to PCM signed 16-bit little-endian, which is the standard uncompressed audio encoding used in WAV files and required by applications expecting raw CD-quality audio data. |
output.wav
|
Defines the output filename with a .wav extension, which causes FFmpeg to wrap the decoded pcm_s16le audio samples in a RIFF/WAV container — the uncompressed audio format developed by Microsoft and IBM. |
Common Use Cases
- Importing audio into a DAW (Digital Audio Workstation) like Audacity, Pro Tools, or Logic Pro, which often works more reliably and accurately with uncompressed PCM WAV files than with lossy MP3s
- Preparing audio for broadcast or podcast mastering pipelines that require uncompressed WAV deliverables conforming to loudness standards like EBU R128
- Loading a music track into DJ software or hardware samplers that only accept WAV or AIFF input formats
- Providing a WAV source file to a CD authoring application, since the Red Book CD audio standard requires uncompressed 16-bit 44.1kHz PCM audio
- Integrating audio into video editing timelines in tools like DaVinci Resolve or Adobe Premiere where WAV is preferred for frame-accurate synchronization and waveform display
- Converting a collection of MP3 archives to WAV before applying audio restoration or noise reduction tools that operate on uncompressed PCM data
Frequently Asked Questions
No. The WAV file produced by this conversion will be uncompressed, but it cannot recover the audio information that was permanently discarded when the original MP3 was encoded. The perceptual artifacts introduced by MP3 compression — such as pre-ringing, frequency smearing, or the Gibbs phenomenon near transients — remain present in the decoded PCM. What you get is an uncompressed representation of the already-compressed audio, not a restoration of the original recording.
MP3 achieves its small file size through lossy compression, typically encoding audio at 128–320 kbps. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression, which for stereo audio at 44.1kHz results in a constant bitrate of 1,411 kbps — roughly 10 MB per minute. A 4 MB MP3 might expand to 40–50 MB as a WAV file, which is expected and correct behavior.
Partially. MP3 files commonly store metadata as ID3 tags, while WAV files use a different metadata scheme called INFO chunks (RIFF metadata). FFmpeg will attempt to map common ID3 fields like title, artist, and album into WAV INFO chunks during conversion, but not all tag fields have direct equivalents and some metadata may be lost or truncated. If preserving rich metadata is critical, consider keeping the original MP3 alongside the WAV.
The flag '-c:a pcm_s16le' instructs FFmpeg to encode the output audio using 16-bit signed PCM in little-endian byte order, which is the standard WAV format compatible with virtually all software and hardware. You can change this to '-c:a pcm_s24le' for 24-bit PCM or '-c:a pcm_s32le' for 32-bit PCM if your downstream application supports it. However, since the source is an MP3, increasing the bit depth beyond 16-bit does not add any real audio information — it only increases file size.
The single-file command shown converts one file at a time. To batch convert all MP3 files in a directory on Linux or macOS, you can use a shell loop: 'for f in *.mp3; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.mp3}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.mp3) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav"'. The online tool processes files individually, so for large batch jobs running FFmpeg locally is recommended.
Yes, WAV with pcm_s16le is one of the most universally accepted formats for audio editing software. DAWs and editors like Audacity, Reaper, and GarageBand all natively support uncompressed WAV without requiring any decoding step during editing, which reduces CPU load and avoids any generation loss from repeated encode/decode cycles. If your DAW supports 32-bit float internally, you might also consider '-c:a pcm_f32le' to match the DAW's internal processing precision.
Technical Notes
The default output codec pcm_s16le produces standard CD-quality uncompressed audio: 16-bit integer samples in little-endian byte order, preserving the sample rate of the source MP3 (most commonly 44100 Hz or 48000 Hz). FFmpeg does not resample or alter the sample rate during this conversion unless explicitly instructed with the '-ar' flag, so the WAV output will have the same sample rate as the MP3 input. The WAV container format has a theoretical file size limit of approximately 4 GB due to its use of 32-bit chunk size fields in the RIFF header — for extremely long audio this can be a constraint, though it is unlikely to be reached with typical MP3 sources under the tool's 1 GB input limit. WAV does not support chapter markers, multiple audio tracks, or embedded subtitles — but neither does MP3, so no such data is lost in this conversion. If the source MP3 was encoded with a variable bitrate (VBR), the decoded PCM output is still uniform fixed-rate PCM; VBR only affects the MP3's internal frame structure, not the decoded audio samples.