Convert FLAC to WAV — Free Online Tool
Convert FLAC lossless audio to WAV format by decoding the FLAC-compressed stream and writing raw PCM audio (16-bit signed little-endian) into a WAV container. The result is fully uncompressed, making it universally compatible with professional audio software, hardware players, and broadcast workflows.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
FLAC stores audio as losslessly compressed PCM data — the raw waveform samples are intact but encoded using a prediction-and-residual algorithm to reduce file size. During conversion, FFmpeg decodes the FLAC stream back to its original uncompressed PCM samples and writes them into a WAV container using the pcm_s16le codec (16-bit signed integer, little-endian byte order). No audio information is lost in either direction: FLAC is lossless going in, and WAV is uncompressed going out. The main change is that WAV carries no compression at all, so the output file will be significantly larger than the FLAC source — typically 2 to 4 times the size, depending on how compressible the original audio was.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the FLAC input, converting the audio, and writing the WAV output. |
-i input.flac
|
Specifies the input file. FFmpeg detects the FLAC container and selects the FLAC decoder to decompress the losslessly encoded PCM audio data back into raw samples. |
-c:a pcm_s16le
|
Sets the audio codec for the output to PCM signed 16-bit little-endian — the standard uncompressed format used in WAV files. This is the same bit depth as CD audio and is universally readable by hardware players, DAWs, and broadcast systems. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to wrap the raw pcm_s16le audio stream in a RIFF/WAV container, which is the structure expected by virtually all audio applications and devices that consume WAV files. |
Common Use Cases
- Preparing audio for import into DAWs like Pro Tools or Logic Pro that require uncompressed WAV files for multi-track editing sessions
- Supplying broadcast-ready masters to radio stations or podcast distributors that mandate uncompressed WAV delivery in 16-bit/44.1kHz or 48kHz format
- Loading music onto older hardware players, DJ controllers, or car stereo systems that support WAV but lack FLAC decoding capability
- Feeding audio into sample editors or hardware samplers that read raw PCM from WAV but cannot parse the FLAC container
- Converting a FLAC music library to WAV before burning audio CDs, since the CD-burning process requires uncompressed PCM and cannot read FLAC directly
- Providing WAV source files to mastering engineers or replication houses whose tools or automated ingest pipelines expect uncompressed audio
Frequently Asked Questions
No — this conversion is entirely lossless in both directions. FLAC is a lossless format that preserves every PCM sample exactly, and WAV stores those same samples uncompressed. The decoded PCM data written into the WAV file is bit-for-bit identical to what the FLAC file encodes. You are not transcoding between different lossy formats, so there is no generation loss whatsoever.
FLAC uses lossless compression to reduce file size — typically achieving 40–60% reduction compared to raw PCM. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression at all. A 30 MB FLAC file might expand to 70–100 MB as a WAV, depending on the audio's complexity and how efficiently FLAC was able to compress it. The audio content is identical; the difference is purely storage overhead.
Partially. FLAC uses Vorbis comment tags (artist, album, title, etc.), while WAV uses a different metadata scheme called INFO chunks or ID3 tags embedded in the RIFF structure. FFmpeg will attempt to map common tags across, but not all DAWs and players read WAV metadata consistently. If tag preservation is critical, verify the output in your target application after conversion. For archival purposes, keeping the original FLAC file alongside the WAV is a good practice.
pcm_s16le stands for PCM, signed 16-bit integer, little-endian byte order — the standard CD-quality audio format and the WAV default. If your FLAC source was encoded at 24-bit depth, you can change the codec to pcm_s24le to preserve the full resolution: ffmpeg -i input.flac -c:a pcm_s24le output.wav. Using 32-bit float is also an option with pcm_f32le, which is common in professional DAW workflows.
Yes. On Linux or macOS you can run: for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.flac}.wav"; done. On Windows Command Prompt: for %f in (*.flac) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This loops through every FLAC file in the current directory and produces a matching WAV file with the same base filename.
Yes — FLAC supports sample rates up to 655,350 Hz in theory, and commonly stores 88.2kHz, 96kHz, or 192kHz hi-res audio. WAV with pcm_s16le (or pcm_s24le for 24-bit) fully supports these sample rates. FFmpeg will automatically carry the source sample rate through to the output without resampling, so a 96kHz/24-bit FLAC will produce a 96kHz/24-bit WAV as long as you specify pcm_s24le instead of the default pcm_s16le.
Technical Notes
The default output codec pcm_s16le targets standard CD-resolution audio (16-bit, 44.1kHz stereo), which is the most universally compatible WAV variant and the format expected by CD authoring tools and most consumer playback software. If your FLAC source was recorded or mastered at 24-bit depth — common for studio masters and hi-res downloads — using pcm_s16le will truncate the extra bits, effectively dithering them away. In that case, substitute -c:a pcm_s24le to preserve the full 24-bit word length. WAV files are limited to 4GB due to the 32-bit file size field in the RIFF header; for very long high-sample-rate recordings that would exceed this, consider WAV64 or RF64 extensions (using FFmpeg's -f wav -rf64 auto flag). FLAC's cue sheet and ReplayGain metadata have no direct WAV equivalents and will be dropped during conversion. The WAV container does not support multiple audio tracks, subtitles, or chapters — none of which FLAC supports either, so no content is lost on those dimensions.