Convert WAV to OPUS Online
The optimal encoding chain: uncompressed WAV directly to Opus. No intermediate loss. Free, in your browser.
.wav · up to 100 MB
What it's for
WAV to OPUS: the perfect encoding chain
No cumulative loss
Starting from uncompressed WAV guarantees the best possible Opus: no artifacts inherited from previous compressions.
Podcasts and web distribution
Opus in OGG is ideal for modern podcast distribution and audio assets in web applications.
100% private
Your studio or microphone recordings are never uploaded to any server. Complete local processing.
Native 48 kHz
Opus operates at 48 kHz, the optimal frequency for voice and music. Automatic high-quality resampling.
How it works
Three steps, no hassle
Upload your WAV file
Drag or select your .wav file. 16 or 24-bit PCM, any sample rate, up to 200 MB.
Direct encoding to Opus
FFmpeg.wasm takes the uncompressed PCM and encodes it directly to Opus with no intermediate decoding step. Maximum possible quality.
Download your OPUS file
The .opus file ready for podcast distribution, web apps, VoIP, or optimized storage.
FAQ
Got questions?
All lossy encoding (like Opus or MP3) introduces artifacts that depend on the input signal. When starting from an already lossy-compressed format (MP3, AAC), the artifacts of the original format compound with the artifacts introduced by the new codec, producing cumulative degradation. Starting from WAV (uncompressed PCM) completely avoids this problem: the Opus encoder receives the most faithful possible audio signal and can apply its perceptual shaping algorithms with all available information. If you have the original recording in WAV (studio, USB microphone, DAW export), converting directly to Opus rather than going through MP3 first produces the best possible Opus at any given bitrate.
For voice podcasts recorded with a microphone: 32 kbps mono is transparent for most listeners and produces files of about 240 KB per minute. 48 kbps mono is the conservative recommendation if you want extra headroom. For podcasts with music or sound effects: 64–80 kbps stereo is the Xiph.org recommendation for practical transparency. Compared to the 128 kbps mono or 192 kbps stereo typically used by MP3 podcasts, Opus files are 3–4 times smaller with equal or better quality. Podcast platforms that support Opus directly (Spotify internally, AntennaPod, Pocket Casts) benefit from the bandwidth reduction.
Opus operates internally at 48 kHz, the sample rate defined in RFC 6716 as the codec's native rate. WAV files generated by DAWs (Pro Tools, Logic Pro, Ableton Live) or recording software are often at 44.1 kHz or 96 kHz. FFmpeg automatically applies a high-quality resample (Kaiser windowed sinc filter) to the input format before Opus encoding. Resampling from 44.1 kHz to 48 kHz is practically transparent. Resampling from 96 kHz to 48 kHz reduces ultrasonic content (above 24 kHz) which is inaudible to the human ear anyway.
WAV files can contain metadata in INFO chunks (title, artist, comment) or in RIFF LIST format. These metadata are transferred to the OGG Opus container as Vorbis Comment tags, the standard metadata format for OGG containers. Typical fields (TITLE, ARTIST, ALBUM, DATE, COMMENT) are preserved. BEXT chunk metadata (Broadcast Wave Format, used in radio and TV recordings) are also recognized by FFmpeg and transferred when possible.
OBS Studio supports Opus as an audio output codec since version 27 (2021). You can configure OBS to record directly to MKV or MP4 with Opus audio. However, if your workflow involves recording in high-quality WAV with OBS or an external recorder and then distributing the audio, converting to Opus here gives you the best control over bitrate and quality. For live streaming, OBS encodes Opus in real time; this tool is for post-processing of existing recordings.
Yes. FFmpeg.wasm supports WAV with PCM bit depths of 8, 16, 24, and 32-bit integers, as well as 32-bit float and 64-bit float IEEE 754 (the export format of many DAWs at 32-bit float). In all cases, FFmpeg internally converts to 32-bit float before passing to the Opus encoder. The higher bit depth of the input WAV does not increase the quality of the resulting Opus beyond the selected bitrate, but it ensures no additional degradation is introduced in the conversion.
Convert WAV to OPUS: direct encoding from uncompressed audio for podcasts and web
WAV (Waveform Audio File Format) is the uncompressed audio container developed by Microsoft and IBM in 1991, based on the RIFF (Resource Interchange File Format) format. A typical WAV file contains PCM (Pulse-Code Modulation) audio with no compression, meaning it represents the audio signal with maximum possible fidelity: each sample is stored as an integer value of 16, 24, or 32 bits directly proportional to the amplitude of the sound wave. One minute of stereo WAV audio at 44.1 kHz and 16 bits takes approximately 10 MB; at 96 kHz and 24 bits, about 34 MB. This uncompressed nature makes WAV the reference format for studio recording, DAW exports (Pro Tools, Logic Pro, Ableton Live, Reaper, FL Studio), recording with professional and semi-professional audio interfaces (Focusrite Scarlett, Universal Audio Apollo, PreSonus AudioBox), and any audio production workflow where maximum fidelity is the priority before any distribution or publication process. The WAV to Opus chain is technically optimal for producing maximum-quality Opus because the libopus encoder receives the audio signal with no prior degradation, no artifacts from previous encodings, with the maximum information available for its perceptual modeling algorithms to work on the most faithful possible representation of the original signal. If you have the original recording in WAV, always convert directly to Opus rather than going through MP3 first, since the intermediate MP3 step introduces irreversible compression artifacts that compound with the Opus codec's own artifacts and permanently degrade the final result. The process is completely free and requires no signup or installation of additional software.
The primary use cases for WAV to Opus conversion are podcast distribution on modern platforms, preparation of audio assets for web applications and games, optimized storage of studio recordings, and audio preparation for VoIP and WebRTC pipelines that use Opus natively. In the podcast production context, the typical production chain involves recording in high-quality WAV with a USB microphone (Blue Yeti, Rode NT-USB, Shure MV7) or audio interface (Focusrite Scarlett, PreSonus AudioBox), editing in Audacity, Adobe Audition, or Descript, and exporting for distribution. Exporting to Opus from WAV produces podcast files 3 to 5 times smaller than MP3 with the same perceived quality, significantly reducing storage costs and distribution bandwidth. For web applications and HTML5 games, Opus audio in OGG or WebM containers loads faster and consumes less bandwidth than MP3 equivalents, improving user experience especially on mobile connections with speed or data limitations. The WAV to Opus chain is recommended by the open-source audio production community precisely because it eliminates every source of cumulative degradation: the Opus encoder receives the most faithful possible signal and can apply its perceptual modeling algorithms with all available information, producing the best possible Opus at any given bitrate. All conversion occurs locally in the user's browser via FFmpeg.wasm, ensuring complete privacy. There is no limit on the number of files per session and no usage restrictions on any available function. The tool is compatible with Chrome, Edge, Firefox, and Safari on desktop and mobile without any additional configuration. The resulting file follows IETF and Xiph.org open standards for maximum compatibility with audio players and software.
Convertir.ai performs the WAV to Opus conversion entirely in the browser with FFmpeg.wasm. The process reads the fmt chunk of the RIFF/WAV container to determine the sample rate, bit depth, and number of channels, decodes the PCM including 8, 16, 24, and 32-bit integer PCM formats and 32 and 64-bit IEEE 754 float (the export format of many DAWs at 32-bit float), applies if necessary a high-quality resample to 48 kHz using FFmpeg's Kaiser windowed sinc filter, which is transparent for any audio content, and encodes with libopus at the selected bitrate. The output container is OGG with Vorbis Comment tags, the standard format for Opus distribution per the Xiph.org specification. For WAV files at 44.1 kHz (the CD standard and most music-oriented DAWs), the resample to 48 kHz introduces an imperceptible pitch difference of less than 0.02 semitones. For WAV files at 48 kHz (the video production and VoIP standard), no resampling is applied. No quantity limit, no signup, no watermark, with complete privacy for studio or professional production recordings processed locally in the browser without any transfer of data to external servers. The process is completely free and requires no signup or installation of additional software. All conversion occurs locally in the user's browser via FFmpeg.wasm, ensuring complete privacy. There is no limit on the number of files per session and no usage restrictions on any available function. The tool is compatible with Chrome, Edge, Firefox, and Safari on desktop and mobile without any additional configuration. The resulting file follows IETF and Xiph.org open standards for maximum compatibility with audio players and software.