Audio to WAV
Audio editors, speech-recognition tools, game engines and analysis software very often insist on uncompressed WAV, because decoding a compressed stream on the fly is unreliable for sample-accurate work. This converts MP3, M4A, OGG or FLAC into standard PCM WAV that those tools will accept. It is free, needs no account, and adds no watermark to your result.
Audio to WAV
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What happens to your file.
FFmpeg decodes the source and writes the resulting samples as PCM WAV, keeping the original sample rate and channel layout so timing and pitch are unchanged. Bit depth follows the source: a 24-bit FLAC or WAV master stays 24-bit rather than being quietly truncated to 16, while a lossy file stays at 16-bit because a decoded MP3 has no greater depth to keep. You can pin a depth instead if a tool demands one. Title and artist tags are carried into the WAV, 5.1 surround keeps its six channels, and a recording longer than a standard WAV can hold (about 4 GB) is written as RF64, the extended WAV that editors read.
Where it stops: Converting a lossy file to WAV does not restore quality: it stores the already-degraded audio without compression, which is why the file becomes far larger while sounding identical to the source. Expect roughly ten times the size of a 128 kbps MP3.
What people use it for.
Everything you need, without a paywall.
Common questions.
Does this make my MP3 sound better?
No. It stops further loss but recovers nothing. The WAV will sound exactly like the MP3 it came from, just much larger.
Why is the WAV so much bigger?
WAV stores every sample uncompressed. A three-minute stereo track at CD quality is about 30 MB regardless of how small the compressed original was.
Is the sample rate changed?
No. Sample rate and channel layout are carried across unchanged, so the audio plays at the correct speed and pitch.
Will my 24-bit master stay 24-bit?
Yes. The depth of a lossless source is read and matched, so a 24-bit FLAC or WAV comes back as 24-bit PCM. Set the bit depth explicitly if you need a fixed one.
You might also need.
Convert audio files between MP3, WAV, M4A, FLAC and OGG using FFmpeg with bitrate and quality controls.
Convert uncompressed WAV audio to compact MP3 with configurable audio quality.
Normalize perceived loudness using FFmpeg loudnorm processing for more consistent playback volume.
Inspect codec, sample rate, channels, duration, bitrate, tags and stream metadata using ffprobe.