OPUS Format Guide
Available Conversions
Convert OPUS to AAC for better device compatibility and streaming services
Convert OPUS to AIFF (Audio Interchange File Format) for professional audio editing and Mac compatibility
Convert OPUS to FLAC lossless format for archival and high-quality audio preservation
Convert OPUS to MP2 (MPEG Audio Layer II) for broadcasting and legacy systems
Convert OPUS to MP3 for universal compatibility across all devices and platforms
Convert OPUS to OGG Vorbis for open-source audio and web streaming
Convert OPUS to uncompressed WAV format for editing and professional production
Convert OPUS to WMA (Windows Media Audio) for Windows ecosystem compatibility
About OPUS Format
Opus is a state-of-the-art, open-source audio codec designed for interactive speech and music transmission over the internet. Standardized by the Internet Engineering Task Force (IETF) in 2012 as RFC 6716, Opus combines the best features of two predecessor codecs: SILK (for speech) and CELT (for music), creating a versatile codec that excels at both. Developed by the Xiph.Org Foundation, Mozilla, and Skype Technologies, Opus is completely free of licensing fees and patents, making it ideal for widespread adoption. The codec is designed for real-time applications with very low latency (as low as 5 milliseconds), making it perfect for VoIP, video conferencing, gaming, and live streaming. Opus delivers excellent audio quality across a wide range of bitrates (from 6 kbps for narrowband speech to 510 kbps for full-bandwidth stereo music), and it can adapt dynamically to changing network conditions. The format supports sample rates from 8 kHz to 48 kHz and handles everything from mono speech to full-bandwidth stereo music and multi-channel audio.
History of OPUS
Opus development began in 2007 when Xiph.Org started work on the CELT codec, while Skype independently developed the SILK codec for its VoIP services. Recognizing the complementary strengths of both codecs—CELT's excellence with music and SILK's efficiency with speech—the IETF's Codec Working Group merged the two technologies in 2010 to create Opus. The collaboration brought together engineers from Xiph.Org Foundation, Mozilla, Skype (later Microsoft), Broadcom, and other organizations. Opus was officially standardized as RFC 6716 in September 2012, with the first stable release of the libopus reference implementation following shortly after. In 2013, Opus became mandatory for WebRTC implementations, cementing its role as the standard codec for real-time web communications. The codec received further updates in 2017 with RFC 8251, adding support for ambisonics and other advanced features. Opus quickly gained adoption due to its technical superiority and royalty-free status: it's now used by WhatsApp, Discord, Facebook Messenger, Zoom, Google Duo, Telegram, YouTube live streaming, Spotify Connect, PlayStation 4 and 5 remote play, and countless other applications. Unlike previous codec standards that took years to gain traction, Opus achieved widespread adoption within just a few years of its release, becoming the de facto standard for internet audio communication.
Key Features and Uses
Opus is a hybrid codec that seamlessly switches between SILK mode (optimized for speech) and CELT mode (optimized for music) based on the audio content, providing optimal quality for all types of audio. The codec supports bitrates from 6 kbps to 510 kbps, making it suitable for everything from low-bandwidth voice calls to high-quality music streaming. Opus delivers extremely low latency, with algorithmic delay as low as 5 milliseconds, making it ideal for real-time two-way communication. The format supports adaptive bitrate, dynamically adjusting quality based on network conditions without interrupting the audio stream. Opus handles packet loss gracefully through forward error correction (FEC) and packet loss concealment, maintaining audio quality even on unreliable networks. The codec supports sample rates from 8 kHz (narrowband) to 48 kHz (fullband) and can encode mono, stereo, and multi-channel audio up to 255 channels for applications like ambisonics and immersive audio. Opus is completely royalty-free and open-source, with reference implementations available under BSD license. The format is designed to be computationally efficient, with low CPU requirements for both encoding and decoding. Opus outperforms all other standardized audio codecs in listening tests across all bitrates and use cases, from speech at 6 kbps to music at 128 kbps.
Common Applications
Opus is the standard codec for WebRTC, powering real-time audio communication in web browsers including Chrome, Firefox, Edge, and Safari. Major communication platforms use Opus exclusively: WhatsApp, Discord, Facebook Messenger, Telegram, Google Meet, Zoom (for certain modes), Microsoft Teams, and Skype all rely on Opus for voice and audio transmission. The codec is used in gaming voice chat systems, including those built into PlayStation 4, PlayStation 5, Nintendo Switch online services, and PC gaming platforms like Steam. YouTube uses Opus for live streaming audio and increasingly for regular video uploads. Music streaming services including Spotify (for Spotify Connect), Deezer, and SoundCloud use Opus for audio delivery. The format is standard in Voice over IP (VoIP) applications, SIP telephony systems, and unified communications platforms. Podcast apps and audio streaming platforms use Opus for efficient, high-quality content delivery. Radio broadcasters employ Opus for internet radio streaming and contribution feeds. Professional audio applications use Opus for low-latency monitoring and remote collaboration. Mobile apps across iOS and Android use Opus for in-app voice and audio features. The codec's combination of superior quality, low latency, adaptability, and royalty-free licensing has made it the clear choice for any application involving real-time audio communication or efficient audio streaming over the internet.
Advantages and Disadvantages
✓ Advantages
- Superior Quality: Best audio quality among all standardized codecs at all bitrates
- Ultra-Low Latency: Algorithmic delay as low as 5 ms, perfect for real-time communication
- Versatile: Excellent for both speech and music in a single codec
- Adaptive: Dynamically adjusts to network conditions and content type
- Wide Bitrate Range: Supports 6 kbps to 510 kbps for all use cases
- Royalty-Free: Completely free of patents and licensing fees
- Open Source: Reference implementation available under BSD license
- Robust: Excellent packet loss concealment and error correction
- Efficient: Low CPU requirements for encoding and decoding
- Modern Standard: Mandatory for WebRTC, widely adopted by major platforms
✗ Disadvantages
- Newer Format: Less support on older devices and legacy systems
- Limited Hardware: Not widely supported by dedicated audio hardware
- iOS Limitations: Limited native support in Apple ecosystem
- Not Universal: Not as ubiquitous as MP3 for casual playback
- Storage Format: Primarily designed for streaming, not archival storage
- File Player Support: Fewer traditional music players support Opus files
- Adoption Lag: Some commercial platforms slow to adopt newer codecs
- Lossy Compression: Cannot match quality of lossless formats like FLAC