{
  "draft": "draft-ietf-codec-opus-16",
  "doc_id": "RFC6716",
  "title": "Definition of the Opus Audio Codec",
  "authors": [
    "JM. Valin",
    "K. Vos",
    "T. Terriberry"
  ],
  "format": [
    "TEXT",
    "HTML"
  ],
  "page_count": "326",
  "pub_status": "PROPOSED STANDARD",
  "status": "PROPOSED STANDARD",
  "source": "Internet Wideband Audio Codec",
  "abstract": "This document defines the Opus interactive speech and audio codec.  Opus is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even live, distributed music performances.  It scales from low bitrate narrowband speech at 6 kbit/s to very high quality stereo music at 510 kbit/s.  Opus uses both Linear Prediction (LP) and the Modified Discrete Cosine Transform (MDCT) to achieve good compression of both speech and music. [STANDARDS-TRACK]",
  "pub_date": "September 2012",
  "keywords": [
    "voice",
    "music",
    "lossy compression",
    "VOIP"
  ],
  "obsoletes": [],
  "obsoleted_by": [],
  "updates": [],
  "updated_by": [
    "RFC8251"
  ],
  "see_also": [],
  "doi": "10.17487/RFC6716",
  "errata_url": "https://www.rfc-editor.org/errata/rfc6716"
}