Opus
Opus is a versatile, open-source audio codec designed for interactive real-time applications like VoIP, video conferencing, and streaming. It provides high-quality audio compression with low latency, supporting a wide range of bitrates from 6 kbps to 510 kbps and sampling rates from 8 kHz to 48 kHz. Developed by the IETF and standardized as RFC 6716, it combines technologies from the SILK and CELT codecs to handle both speech and music efficiently.
Developers should learn Opus for applications requiring real-time audio communication, such as webRTC-based video calls, online gaming voice chat, or live streaming platforms, due to its low latency and adaptive bitrate capabilities. It's also essential for building audio-focused tools like podcast editors or music streaming services where high compression efficiency and quality are critical, as it outperforms older codecs like MP3 and AAC in many scenarios.