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OPUS
Low-latency audio codec for speech and music
BSD-3-Clause
ABOUT
Real-time speech and music over the network need a codec that is both low-latency and high-quality. Proprietary options lock products into licenses. Opus is an IETF/Xiph codec so calls, streams, and WebRTC encode audio without a closed AMR or AAC stack.
INSTALL
brew install opusINTEGRATION GUIDE
1. Encode voice for WebRTC or VoIP at low bitrate and latency
2. Stream music or podcasts with an open codec instead of AAC
3. Decode Opus in a player, game, or conferencing stack
4. Package reference audio for ML speech datasets in an open format
TAGS
opusaudiocodecwebrtcspeechxiphstreaming