Audio Converter

Convert WAV to the Codec Powering Discord, YouTube, and Every Modern Voice App

Opus is the IETF internet standard for audio — the codec inside Discord calls, YouTube streams, WhatsApp voice messages, and every WebRTC-based application. It delivers better quality than MP3 and AAC at the same bitrate, and dramatically outperforms both at low bitrates. Upload your WAV and get an Opus file built for modern internet delivery.

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Convert WAV to the Codec Powering Discord, YouTube, and Every Modern Voice App

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Audio Format Converter

Upload any audio file, choose an output format, and download the converted result. Supports MP3, WAV, FLAC, AAC, OGG, ALAC, WMA, Opus and M4A.

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Person streaming audio on a mobile device over a cellular connection

Unmatched Quality at Low Bitrates — Better Than MP3 and AAC

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Unmatched Quality at Low Bitrates — Better Than MP3 and AAC

Opus at 64 kbps outperforms MP3 at 128 kbps and AAC at 96 kbps in listening tests for music content. At very low bitrates (16-32 kbps), the gap is even more dramatic for speech. Opus achieves this by combining two codec technologies in a single algorithm: SILK for speech-optimised encoding and CELT for music and general audio. The encoder switches between them automatically depending on the content.

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People on a voice and video call across different devices

The Codec Behind Discord, YouTube, WhatsApp, and WebRTC

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The Codec Behind Discord, YouTube, WhatsApp, and WebRTC

Discord encodes all voice and audio in Opus. YouTube uses Opus for audio in WebM streams. WhatsApp, Signal, Telegram, Zoom, and Google Meet all use Opus for voice transmission via WebRTC. The Web Audio API and the MediaRecorder API in all major browsers produce Opus by default when recording audio. Converting to Opus means your audio is in the format the entire modern internet audio stack was built around.

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Podcaster or streamer recording live audio in a home studio

Designed for Streaming: Ultra-Low Latency at 5-10 ms Frames

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Designed for Streaming: Ultra-Low Latency at 5-10 ms Frames

Opus was engineered specifically for real-time audio transmission over the internet. It operates at frame sizes as small as 2.5 ms, with typical voice call configurations using 10-20 ms frames. This makes Opus the only codec suitable for both live streaming and stored audio — using the same encoded format whether you are broadcasting a live stream or serving a pre-recorded podcast.

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How It Works

Three steps from WAV to a streaming-optimised Opus file

Upload your WAV file

Drop any WAV file onto the tool or click to browse. Voice recordings, music tracks, podcast audio, and sound effects are all accepted.

Convert to Opus

Click "Convert". FFmpeg encodes the PCM data using the libopus encoder and wraps it in an Ogg container, producing a standard .opus file.

Download your Opus file

Preview the output, then download your .opus file ready for web delivery, streaming servers, Discord bots, or any WebRTC-based application.

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Deep Dive

WAV vs Opus: Uncompressed Archive vs the Internet's Audio Codec

WAV stores raw PCM for perfect fidelity in the studio. Opus was designed by the IETF to be the best possible audio codec for internet transmission — combining speech and music encoding in a single, patent-free algorithm.

Voice call users, streamers, podcasters, and developers using Opus audio

WAV: Uncompressed PCM, Ideal for Production

WAV stores audio as raw linear PCM — every sample from the original recording written to disk without compression. This makes WAV perfect for recording, editing, and mastering inside a DAW, where every bit of the original signal must be preserved for processing. Its 10 MB per minute footprint makes it impractical for internet delivery, real-time streaming, or mobile applications.

TypeUncompressed PCM in a RIFF container (lossless)

Size~10 MB per minute at 44.1 kHz / 16-bit stereo

LatencyNo codec latency, but large packet sizes for streaming

Keep WAV for studio production and archiving. Convert to Opus for any internet delivery scenario.

Opus: The IETF Internet Audio Standard

Opus (RFC 6716) was standardised by the Internet Engineering Task Force in 2012 as the recommended audio codec for real-time communications over IP. It combines two encoding technologies: SILK, the speech-optimised codec from Skype, and CELT, a low-latency music codec from Xiph.Org. The encoder selects the optimal mode for each 20 ms frame based on whether the content is speech, music, or a mix, producing the best possible quality at any bitrate from 6 kbps to 510 kbps.

StandardIETF RFC 6716 — the official internet audio codec standard

Bitrate range6 kbps (low-quality voice) to 510 kbps (transparent music)

Frame sizes2.5, 5, 10, 20, 40, or 60 ms — configurable per use case

PatentsNone — completely patent-free and royalty-free

Quality at Low Bitrates: Where Opus Dominates

At 64 kbps, Opus produces audio that blind listening tests rate as equivalent or superior to MP3 at 128 kbps and AAC at 96 kbps. At 32 kbps, Opus delivers good music quality where MP3 and AAC already show noticeable artefacts. For speech at 16-24 kbps, Opus maintains near-telephone-quality intelligibility where older codecs break down entirely. This low-bitrate efficiency is the primary reason voice communication apps universally chose Opus.

Speech quality16-32 kbps covers voice calls with high intelligibility

Music quality64 kbps beats MP3 at 128 kbps in listening tests

Transparent music128-192 kbps considered transparent for most content

For stored audio files (not live streaming), 96 kbps Opus is considered excellent quality for music by most listeners.

Browser Support: Opus in Web Audio API and MediaRecorder

All major browsers (Chrome, Firefox, Edge, Opera, and Safari since version 16) support Opus playback through the HTML5 `<audio>` element and the Web Audio API. The MediaRecorder API records audio in Opus by default in Chrome and Firefox. WebRTC, the technology behind in-browser voice and video calls, mandates Opus as a required codec in the specification — every WebRTC implementation must support it.

HTML5 AudioChrome, Firefox, Edge, Opera, Safari 16+

MediaRecorder defaultOpus in Chrome and Firefox by default

WebRTCMandated by the WebRTC specification — universally supported

Our converter uses FFmpeg's libopus encoder at a high bitrate to produce Opus files suitable for both streaming delivery and stored audio.

Benefits

Why Convert WAV to Opus

📡

Best Quality Per Kilobit

Opus at 64 kbps outperforms MP3 at 128 kbps. At low bitrates, the gap versus older codecs is even larger — no other widely deployed codec matches Opus below 96 kbps.

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Powers Discord, YouTube, WebRTC

Discord, YouTube WebM streams, WhatsApp, Signal, Zoom, and every WebRTC application encode audio in Opus. Your file is in the same format the internet audio stack is built around.

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Ultra-Low Latency

Opus frame sizes start at 2.5 ms, enabling real-time streaming with codec latency an order of magnitude lower than MP3 or AAC.

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Patent-Free IETF Standard

Opus is an IETF standard with no patents and no royalties — safe to use in any commercial or open-source project without licensing concerns.

Who Uses It

Who Converts WAV to Opus

Discord bot developers, web audio engineers, podcast producers targeting modern platforms, WebRTC app developers, and streamers optimising audio bandwidth.

Discord users, WebRTC developers, streamers, and podcasters using Opus audio

Discord Bot Developers

Discord's audio pipeline encodes all audio in Opus. Discord bots that play music, sound effects, or voice responses must send Opus-encoded audio frames to the voice channel. Pre-converting WAV assets to Opus eliminates the real-time transcoding step, reducing CPU load on the bot server and improving audio start latency.

Discord bitrate64 kbps Opus for normal quality, 96 kbps for high quality

WebRTC and Real-Time Communication Developers

The WebRTC specification mandates Opus support in all browser implementations. Developers building video conferencing tools, voice chat features, or real-time collaboration apps using WebRTC work with Opus natively. Converting audio assets, prompts, or pre-recorded announcements to Opus aligns them with the format the WebRTC audio pipeline handles most efficiently.

Web Audio and Streaming Engineers

Web apps that serve audio over HTTP use Opus inside WebM containers for the best quality-to-bandwidth ratio available in a browser-native format. At 96-128 kbps Opus, music streaming quality is excellent while consuming less than half the bandwidth of an equivalent AAC stream at 256 kbps.

Podcast Producers Targeting Modern Platforms

A growing number of podcast platforms and RSS aggregators accept Opus as an audio format. Opus at 96 kbps produces near-transparent speech quality at roughly half the file size of a standard 192 kbps MP3 episode — meaningful for podcast hosts charging by storage or bandwidth.

Live Streamers Optimising Audio Bandwidth

Streamers who encode their own audio for custom ingest pipelines or self-hosted streaming servers can use Opus to reduce audio bitrate without the quality penalty that comes with MP3 or AAC at the same settings. Opus at 64 kbps leaves more of the available upload bandwidth for video.

Mobile App Developers

Android and iOS both support Opus playback. Mobile apps that bundle audio content — voice responses, notification sounds, or music clips — benefit from Opus's superior low-bitrate quality. Smaller audio files also reduce app download size and in-app streaming bandwidth.

Converts using FFmpeg on dedicated servers.

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Frequently Asked Questions

What makes Opus different from OGG Vorbis?

OGG Vorbis and Opus are both open-source and patent-free, but they are different codecs. Opus was designed specifically for internet transmission and combines two encoding technologies — SILK for speech and CELT for music — in a single algorithm. Opus consistently outperforms OGG Vorbis at low bitrates (below 128 kbps) and was standardised by the IETF as the recommended internet audio codec. OGG Vorbis is older and better suited for game assets and offline storage; Opus is better for streaming and real-time communication.

Does Opus play in all browsers?

Yes. Chrome, Firefox, Edge, Opera, and Safari (since version 16) all support Opus playback natively. Opus inside a WebM container is the preferred format for the Web Audio API and is the default output of the MediaRecorder API in Chrome and Firefox. The WebRTC specification also mandates Opus support in all browser implementations.

Why do Discord bots need Opus?

Discord's voice pipeline transmits audio as Opus-encoded frames over the Discord voice protocol. Bots that play audio in voice channels must send Opus frames directly. If a bot starts with a WAV or MP3 file, it must encode to Opus in real time. Pre-converting assets to Opus eliminates this real-time transcoding step and reduces CPU usage on the bot server.

Does converting WAV to Opus reduce quality?

Yes — Opus is a lossy codec. The conversion discards audio information the ear typically cannot perceive at the chosen bitrate. At 96-128 kbps, most listeners cannot distinguish Opus from the original WAV in a blind test. At 64 kbps, quality is still excellent for music — comparable to MP3 at 128 kbps. Keep the original WAV as your master archive.

What bitrate should I use for Opus?

For speech (podcasts, voice messages): 16-32 kbps is sufficient for high intelligibility; 48-64 kbps is excellent. For music: 64 kbps is good, 96 kbps is very good, 128 kbps is considered transparent by most listeners. For archiving, 192 kbps Opus is near-indistinguishable from lossless for typical content.

Is there a file size limit?

The upload limit is 100 MB per file. Most WAV audio files are well within this limit.

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