Convert OGG to WAV — Free Online Tool
Convert OGG audio files (Vorbis, Opus, or FLAC streams) to WAV format using uncompressed PCM audio, producing a universally compatible audio file ideal for professional workflows, audio editing, and archiving. This conversion fully decodes the OGG container's lossy or lossless stream into raw 16-bit PCM samples — the standard uncompressed format understood by virtually every audio tool ever made.
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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
OGG files most commonly contain Vorbis or Opus audio, both of which are lossy compressed codecs. During this conversion, FFmpeg decodes the compressed audio stream — decompressing every frame from its lossy representation back into raw PCM samples — and then writes those samples into a WAV container using the pcm_s16le codec (signed 16-bit little-endian PCM). This is a full decode, not a remux: the audio is reconstructed from the compressed data, quantized to 16-bit depth, and stored without any further compression. If the source OGG contained a FLAC stream (losslessly compressed), the output WAV will be a bit-accurate reconstruction of the original PCM data. For Vorbis or Opus sources, however, the quality ceiling is permanently set by the original lossy encoding — decoding to WAV does not recover information discarded during that initial compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all decoding and encoding. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — your OGG file never leaves your machine. |
-i input.ogg
|
Specifies the input file — an OGG container that may hold a Vorbis, Opus, or FLAC audio stream. FFmpeg auto-detects the codec inside the OGG wrapper and selects the appropriate decoder without any additional flags. |
-c:a pcm_s16le
|
Sets the audio encoder to signed 16-bit little-endian PCM, the standard uncompressed audio format used in WAV files. This fully decodes the compressed OGG audio stream and re-encodes it as raw PCM samples — the format required by the WAV container and understood universally by audio software and hardware. |
output.wav
|
Specifies the output filename with a .wav extension, which tells FFmpeg to wrap the pcm_s16le audio stream in a RIFF/WAV container — the standard uncompressed audio file format developed by Microsoft and IBM, with near-universal compatibility across operating systems, DAWs, and broadcast systems. |
Common Use Cases
- Importing OGG game audio assets (common in open-source games like those using the Godot engine) into a DAW like Audacity or Reaper, which may have better native support for WAV than OGG
- Preparing OGG podcast recordings or voice memos for submission to broadcast platforms or radio stations that require uncompressed WAV files for quality assurance
- Converting OGG music or sound effects into WAV so they can be used in video editing software like DaVinci Resolve or Adobe Premiere that works more reliably with uncompressed audio on the timeline
- Archiving an OGG audio collection to uncompressed WAV as a long-term preservation format, eliminating dependency on the Vorbis or Opus decoder for future playback
- Preparing OGG-format vocal or instrument recordings for mastering engineers who require flat, uncompressed WAV delivery files
- Converting OGG sound samples from a free sound library into WAV format for use in hardware samplers or older audio hardware that only reads PCM WAV files
Frequently Asked Questions
No — and this is the most important thing to understand about this conversion. Vorbis and Opus are lossy codecs, meaning they permanently discard audio information during the original OGG encoding. Decoding to WAV simply unpacks what survived that compression into uncompressed PCM; it cannot reconstruct lost data. The resulting WAV will sound identical to the OGG source but will be much larger. The only exception is if your OGG file contains a FLAC stream, which is lossless — in that case, the WAV will be a perfect reconstruction of the original PCM audio.
OGG/Vorbis typically achieves compression ratios of 8:1 to 12:1 compared to uncompressed audio. A 5MB OGG file could easily become 40–80MB as a WAV, because WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression whatsoever. The file size of the WAV is determined entirely by duration, sample rate, and channel count — a stereo 44.1kHz WAV uses exactly 176,400 bytes per second regardless of the audio content.
Partially — WAV supports a limited form of metadata through its INFO chunk (part of the RIFF specification), and FFmpeg will attempt to map common OGG Vorbis comment fields like TITLE, ARTIST, and ALBUM into WAV INFO tags. However, WAV's metadata support is far less robust than OGG's, and some tags may be dropped or truncated. OGG chapter markers, which the format supports natively, are not preserved in WAV since WAV has no chapter structure. If metadata preservation is critical, consider using a dedicated metadata editor after conversion.
Yes — you can modify the FFmpeg command by changing the audio codec flag. Replace -c:a pcm_s16le with -c:a pcm_s24le for 24-bit output, or -c:a pcm_s32le for 32-bit. However, if your source is an OGG/Vorbis or OGG/Opus file (both lossy), the extra bit depth will not add real audio information — the dynamic range ceiling is already constrained by the lossy encoding. Higher bit depth is most meaningful when the OGG source contains a FLAC (lossless) stream.
On Linux or macOS, you can run a shell loop in your terminal: for f in *.ogg; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.ogg}.wav"; done. On Windows Command Prompt, use: for %f in (*.ogg) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This applies the same pcm_s16le codec used by this tool to every OGG file in the current directory, outputting a matching WAV file for each. This is especially useful for processing large collections of game audio assets or sample libraries.
FFmpeg handles both OGG/Vorbis and OGG/Opus seamlessly with the same command — it auto-detects the codec in the container and decodes accordingly. Opus audio will be fully decoded to PCM and written to WAV just like Vorbis. Note that Opus is optimized for speech and lower bitrates, so if your OGG/Opus file was encoded at a low bitrate (under 64 kbps), the WAV output may reveal compression artifacts that were less noticeable in the compressed stream.
Technical Notes
This conversion targets the most universally compatible WAV subformat: pcm_s16le, which is signed 16-bit PCM in little-endian byte order. This is the standard used by CD audio and is readable by every DAW, audio editor, sampler, broadcast system, and operating system audio API without any additional codec installation. The sample rate and channel layout from the OGG source are preserved by default — a 48kHz stereo OGG/Opus file will produce a 48kHz stereo WAV, while a 44.1kHz OGG/Vorbis file will produce a 44.1kHz WAV. WAV files using pcm_s16le are limited to a maximum file size of approximately 4GB due to the RIFF/WAV format's 32-bit size fields, which corresponds to roughly 6.7 hours of stereo 44.1kHz audio. OGG's multi-track audio support and chapter markers are not carried over to WAV, as WAV supports only a single audio stream and has no chapter structure. If your workflow requires those features, consider WAV64 or another container. The OGG container's streaming and open-format characteristics are irrelevant once decoded to WAV.