Convert WAV to WEBA — Free Online Tool
Convert WAV audio files to WEBA format using the Opus codec, shrinking uncompressed PCM audio into a highly efficient, web-optimized stream. WEBA with Opus delivers near-transparent quality at 128 kbps — a fraction of the size of a WAV file — making it ideal for serving audio on the web.
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FFmpeg Command
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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
WAV files typically store audio as uncompressed PCM (most commonly 16-bit signed little-endian, pcm_s16le), which preserves every sample exactly but results in very large files. During this conversion, FFmpeg decodes the raw PCM samples from the WAV container and re-encodes them using the Opus codec (libopus), a modern lossy audio codec developed by the IETF. The encoded Opus stream is then wrapped in a WEBA container, which is an audio-only variant of the WebM format. Because Opus is a lossy codec, some audio data is discarded during encoding — the algorithm uses psychoacoustic modeling to remove frequencies and details less perceptible to human hearing. The result is a file that is typically 5 to 15 times smaller than the original WAV, depending on bitrate and source material.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles the full pipeline: reading the WAV container, decoding the PCM audio, resampling it to 48 kHz for Opus compatibility, encoding with libopus, and writing the WEBA output. |
-i input.wav
|
Specifies the input WAV file. FFmpeg reads the WAV container header to determine the audio codec (typically pcm_s16le), sample rate, channel count, and bit depth before beginning decoding. |
-c:a libopus
|
Selects the Opus encoder (libopus) for the audio stream. Opus is the preferred codec for WEBA files, offering better compression efficiency than Vorbis at all bitrates, particularly for speech and mixed content below 128 kbps. |
-b:a 128k
|
Sets the target audio bitrate to 128 kilobits per second for the Opus stream. At this bitrate, Opus produces audio that is perceptually transparent for most music and speech content, while reducing the file size by approximately 10–11x compared to a 44.1 kHz / 16-bit WAV source. |
-vn
|
Explicitly disables any video stream in the output. This flag is included as a safeguard because WAV files should not contain video, but it ensures the WEBA file is strictly audio-only as the format requires. |
output.weba
|
Specifies the output filename and container. The .weba extension signals an audio-only WebM container; FFmpeg uses the extension to select the WebM muxer and writes the Opus-encoded audio stream into it. |
Common Use Cases
- Preparing audio assets for a web application or HTML5 audio player, where WAV files would be too large to stream efficiently over typical internet connections
- Converting a WAV recording of a podcast or voiceover to WEBA/Opus for embedding on a website, since Opus handles speech content exceptionally well even at lower bitrates like 64k or 96k
- Reducing the file size of a WAV sound effects library for a browser-based game, where fast load times are critical and lossless quality is not required
- Transcoding WAV audio exported from a DAW into a web-compatible format for delivery via a streaming platform or CDN that requires WebM/Opus
- Converting uncompressed WAV recordings captured by broadcast or field equipment into a compressed format suitable for web distribution without first re-importing into an NLE
- Creating a web-compatible version of a WAV audio file for use in a Progressive Web App (PWA) or web-based audio tool, where WEBA/Opus is natively supported in Chromium-based browsers and Firefox
Frequently Asked Questions
The size reduction depends on the WAV file's sample rate and bit depth, plus the target bitrate chosen for Opus. A standard stereo WAV at 44.1 kHz / 16-bit runs at roughly 1,411 kbps uncompressed. Encoding to Opus at the default 128 kbps yields an approximately 11x reduction in file size. For a 50 MB WAV file, the resulting WEBA file would typically be around 4–5 MB at 128k.
Yes — Opus is a lossy codec, so some audio information is permanently discarded during encoding. At 128 kbps, most listeners find the quality indistinguishable from the uncompressed source for music and speech. For critical listening or archival purposes, the original WAV should be kept. If the WAV source contains high-resolution audio (24-bit or 32-bit PCM), the down-conversion to Opus's internal processing depth means that extra resolution will not be preserved in the output.
WEBA supports two audio codecs: Opus (libopus) and Vorbis (libvorbis). Opus is the modern successor to Vorbis and outperforms it at virtually every bitrate, especially below 128 kbps. Opus also has significantly lower latency, which matters for interactive web applications. This tool defaults to libopus because it is the recommended codec for new WEBA files by the WebM project and the IETF.
Replace the value after the -b:a flag with your desired bitrate. For example, use -b:a 64k for smaller speech-only files, -b:a 192k for higher-fidelity music, or -b:a 320k for the highest quality Opus encoding. The full command would look like: ffmpeg -i input.wav -c:a libopus -b:a 192k -vn output.weba. Note that Opus is highly efficient, and most listeners cannot distinguish 128k Opus from lossless audio for typical content.
Basic metadata embedded in the WAV file's INFO chunk or ID3 tags (such as title, artist, and album) may be carried over by FFmpeg into the WebM container's Matroska-style tag structure. However, WAV metadata support is inconsistent across software, and some tags may not survive the conversion cleanly. If metadata preservation is critical, verify the output tags with a tool like MediaInfo or ffprobe after conversion.
Yes. On Linux or macOS, you can run: for f in *.wav; do ffmpeg -i "$f" -c:a libopus -b:a 128k -vn "${f%.wav}.weba"; done. On Windows Command Prompt, use: for %f in (*.wav) do ffmpeg -i "%f" -c:a libopus -b:a 128k -vn "%~nf.weba". This applies the same Opus encoding settings to every WAV file in the current directory, which is particularly useful when converting a large sound library for web deployment.
Technical Notes
WAV files may contain audio encoded in several codecs beyond plain PCM — including ADPCM, A-law, mu-law, or even FLAC — but the vast majority of WAV files encountered in practice use 16-bit signed little-endian PCM (pcm_s16le). FFmpeg handles all of these transparently when decoding to feed into the Opus encoder. The libopus encoder internally operates at 48 kHz, so if your WAV source is at 44.1 kHz or another sample rate, FFmpeg will automatically resample the audio to 48 kHz before encoding — this is normal and expected behavior for Opus. WEBA is an audio-only WebM container, meaning it uses the same .webm-derived structure but the -vn flag ensures no video stream is written. Browser compatibility for WEBA/Opus is excellent in Chrome, Firefox, and Edge, but Safari added Opus support only in version 11 (macOS) and iOS 16 — older Apple devices and Safari versions will not play WEBA files natively. For maximum cross-browser reach, consider also providing an MP3 or AAC fallback. The WEBA format does not support chapters, embedded subtitles, or multiple audio tracks.