1873 lines
77 KiB
XML
Executable file
1873 lines
77 KiB
XML
Executable file
<?xml version="1.0" encoding="utf-8"?>
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<!--
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Copyright (c) 2012-2016 Xiph.Org Foundation and contributors
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions
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are met:
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- Redistributions of source code must retain the above copyright
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notice, this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
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``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
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LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
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A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
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OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
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EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
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PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
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LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
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NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
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SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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Special permission is granted to remove the above copyright notice, list of
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conditions, and disclaimer when submitting this document, with or without
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modification, to the IETF.
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-->
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<!DOCTYPE rfc SYSTEM 'rfc2629.dtd' [
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<!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'>
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<!ENTITY rfc3533 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3533.xml'>
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<!ENTITY rfc3629 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3629.xml'>
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<!ENTITY rfc4732 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4732.xml'>
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<!ENTITY rfc5226 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5226.xml'>
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<!ENTITY rfc5334 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5334.xml'>
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<!ENTITY rfc6381 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6381.xml'>
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<!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'>
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<!ENTITY rfc6982 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6982.xml'>
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<!ENTITY rfc7587 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.7587.xml'>
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]>
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<?rfc toc="yes" symrefs="yes" ?>
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<rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-14"
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updates="5334">
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<front>
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<title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title>
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<author initials="T.B." surname="Terriberry" fullname="Timothy B. Terriberry">
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<organization>Mozilla Corporation</organization>
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<address>
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<postal>
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<street>650 Castro Street</street>
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<city>Mountain View</city>
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<region>CA</region>
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<code>94041</code>
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<country>USA</country>
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</postal>
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<phone>+1 650 903-0800</phone>
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<email>tterribe@xiph.org</email>
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</address>
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</author>
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<author initials="R." surname="Lee" fullname="Ron Lee">
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<organization>Voicetronix</organization>
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<address>
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<postal>
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<street>246 Pulteney Street, Level 1</street>
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<city>Adelaide</city>
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<region>SA</region>
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<code>5000</code>
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<country>Australia</country>
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</postal>
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<phone>+61 8 8232 9112</phone>
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<email>ron@debian.org</email>
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</address>
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</author>
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<author initials="R." surname="Giles" fullname="Ralph Giles">
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<organization>Mozilla Corporation</organization>
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<address>
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<postal>
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<street>163 West Hastings Street</street>
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<city>Vancouver</city>
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<region>BC</region>
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<code>V6B 1H5</code>
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<country>Canada</country>
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</postal>
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<phone>+1 778 785 1540</phone>
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<email>giles@xiph.org</email>
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</address>
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</author>
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<date day="22" month="February" year="2016"/>
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<area>RAI</area>
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<workgroup>codec</workgroup>
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<abstract>
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<t>
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This document defines the Ogg encapsulation for the Opus interactive speech and
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audio codec.
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This allows data encoded in the Opus format to be stored in an Ogg logical
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bitstream.
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</t>
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</abstract>
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</front>
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<middle>
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<section anchor="intro" title="Introduction">
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<t>
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The IETF Opus codec is a low-latency audio codec optimized for both voice and
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general-purpose audio.
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See <xref target="RFC6716"/> for technical details.
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This document defines the encapsulation of Opus in a continuous, logical Ogg
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bitstream <xref target="RFC3533"/>.
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Ogg encapsulation provides Opus with a long-term storage format supporting
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all of the essential features, including metadata, fast and accurate seeking,
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corruption detection, recapture after errors, low overhead, and the ability to
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multiplex Opus with other codecs (including video) with minimal buffering.
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It also provides a live streamable format, capable of delivery over a reliable
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stream-oriented transport, without requiring all the data, or even the total
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length of the data, up-front, in a form that is identical to the on-disk
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storage format.
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</t>
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<t>
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Ogg bitstreams are made up of a series of 'pages', each of which contains data
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from one or more 'packets'.
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Pages are the fundamental unit of multiplexing in an Ogg stream.
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Each page is associated with a particular logical stream and contains a capture
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pattern and checksum, flags to mark the beginning and end of the logical
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stream, and a 'granule position' that represents an absolute position in the
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stream, to aid seeking.
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A single page can contain up to 65,025 octets of packet data from up to 255
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different packets.
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Packets can be split arbitrarily across pages, and continued from one page to
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the next (allowing packets much larger than would fit on a single page).
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Each page contains 'lacing values' that indicate how the data is partitioned
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into packets, allowing a demultiplexer (demuxer) to recover the packet
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boundaries without examining the encoded data.
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A packet is said to 'complete' on a page when the page contains the final
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lacing value corresponding to that packet.
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</t>
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<t>
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This encapsulation defines the contents of the packet data, including
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the necessary headers, the organization of those packets into a logical
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stream, and the interpretation of the codec-specific granule position field.
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It does not attempt to describe or specify the existing Ogg container format.
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Readers unfamiliar with the basic concepts mentioned above are encouraged to
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review the details in <xref target="RFC3533"/>.
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</t>
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</section>
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<section anchor="terminology" title="Terminology">
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<t>
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The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
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"SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this
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document are to be interpreted as described in <xref target="RFC2119"/>.
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</t>
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</section>
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<section anchor="packet_organization" title="Packet Organization">
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<t>
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An Ogg Opus stream is organized as follows (see
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<xref target="packet-org-example"/> for an example).
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</t>
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<figure anchor="packet-org-example"
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title="Example packet organization for a logical Ogg Opus stream"
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align="center">
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<artwork align="center"><![CDATA[
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Page 0 Pages 1 ... n Pages (n+1) ...
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+------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
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| | | | | | | | | | | | |
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|+----------+| |+-----------------+| |+-------------------+ +-----
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|||ID Header|| || Comment Header || ||Audio Data Packet 1| | ...
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|+----------+| |+-----------------+| |+-------------------+ +-----
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| | | | | | | | | | | | |
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+------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
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^ ^ ^
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| | |
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| | Mandatory Page Break
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| |
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| ID header is contained on a single page
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'Beginning Of Stream'
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]]></artwork>
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</figure>
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<t>
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There are two mandatory header packets.
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The first packet in the logical Ogg bitstream MUST contain the identification
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(ID) header, which uniquely identifies a stream as Opus audio.
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The format of this header is defined in <xref target="id_header"/>.
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It is placed alone (without any other packet data) on the first page of
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the logical Ogg bitstream, and completes on that page.
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This page has its 'beginning of stream' flag set.
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</t>
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<t>
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The second packet in the logical Ogg bitstream MUST contain the comment header,
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which contains user-supplied metadata.
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The format of this header is defined in <xref target="comment_header"/>.
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It MAY span multiple pages, beginning on the second page of the logical
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stream.
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However many pages it spans, the comment header packet MUST finish the page on
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which it completes.
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</t>
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<t>
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All subsequent pages are audio data pages, and the Ogg packets they contain are
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audio data packets.
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Each audio data packet contains one Opus packet for each of N different
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streams, where N is typically one for mono or stereo, but MAY be greater than
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one for multichannel audio.
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The value N is specified in the ID header (see
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<xref target="channel_mapping"/>), and is fixed over the entire length of the
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logical Ogg bitstream.
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</t>
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<t>
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The first (N - 1) Opus packets, if any, are packed one after another
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into the Ogg packet, using the self-delimiting framing from Appendix B of
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<xref target="RFC6716"/>.
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The remaining Opus packet is packed at the end of the Ogg packet using the
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regular, undelimited framing from Section 3 of <xref target="RFC6716"/>.
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All of the Opus packets in a single Ogg packet MUST be constrained to have the
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same duration.
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An implementation of this specification SHOULD treat any Opus packet whose
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duration is different from that of the first Opus packet in an Ogg packet as
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if it were a malformed Opus packet with an invalid Table Of Contents (TOC)
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sequence.
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</t>
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<t>
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The TOC sequence at the beginning of each Opus packet indicates the coding
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mode, audio bandwidth, channel count, duration (frame size), and number of
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frames per packet, as described in Section 3.1
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of <xref target="RFC6716"/>.
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The coding mode is one of SILK, Hybrid, or Constrained Energy Lapped Transform
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(CELT).
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The combination of coding mode, audio bandwidth, and frame size is referred to
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as the configuration of an Opus packet.
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</t>
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<t>
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Packets are placed into Ogg pages in order until the end of stream.
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Audio data packets might span page boundaries.
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The first audio data page could have the 'continued packet' flag set
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(indicating the first audio data packet is continued from a previous page) if,
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for example, it was a live stream joined mid-broadcast, with the headers
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pasted on the front.
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If a page has the 'continued packet' flag set and one of the following
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conditions is also true:
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<list style="symbols">
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<t>the previous page with packet data does not end in a continued packet (does
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not end with a lacing value of 255) OR</t>
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<t>the page sequence numbers are not consecutive,</t>
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</list>
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then a demuxer MUST NOT attempt to decode the data for the first packet on the
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page unless the demuxer has some special knowledge that would allow it to
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interpret this data despite the missing pieces.
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An implementation MUST treat a zero-octet audio data packet as if it were a
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malformed Opus packet as described in
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Section 3.4 of <xref target="RFC6716"/>.
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</t>
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<t>
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A logical stream ends with a page with the 'end of stream' flag set, but
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implementations need to be prepared to deal with truncated streams that do not
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have a page marked 'end of stream'.
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There is no reason for the final packet on the last page to be a continued
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packet, i.e., for the final lacing value to be 255.
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However, demuxers might encounter such streams, possibly as the result of a
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transfer that did not complete or of corruption.
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If a packet continues onto a subsequent page (i.e., when the page ends with a
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lacing value of 255) and one of the following conditions is also true:
|
|
<list style="symbols">
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<t>the next page with packet data does not have the 'continued packet' flag
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set OR</t>
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<t>there is no next page with packet data OR</t>
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<t>the page sequence numbers are not consecutive,</t>
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</list>
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then a demuxer MUST NOT attempt to decode the data from that packet unless the
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demuxer has some special knowledge that would allow it to interpret this data
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despite the missing pieces.
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There MUST NOT be any more pages in an Opus logical bitstream after a page
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marked 'end of stream'.
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</t>
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</section>
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<section anchor="granpos" title="Granule Position">
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<t>
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The granule position MUST be zero for the ID header page and the
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page where the comment header completes.
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That is, the first page in the logical stream, and the last header
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page before the first audio data page both have a granule position of zero.
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</t>
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<t>
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The granule position of an audio data page encodes the total number of PCM
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samples in the stream up to and including the last fully-decodable sample from
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the last packet completed on that page.
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The granule position of the first audio data page will usually be larger than
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zero, as described in <xref target="start_granpos_restrictions"/>.
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</t>
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<t>
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A page that is entirely spanned by a single packet (that completes on a
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subsequent page) has no granule position, and the granule position field is
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set to the special value '-1' in two's complement.
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</t>
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<t>
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The granule position of an audio data page is in units of PCM audio samples at
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a fixed rate of 48 kHz (per channel; a stereo stream's granule position
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does not increment at twice the speed of a mono stream).
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It is possible to run an Opus decoder at other sampling rates,
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but all Opus packets encode samples at a sampling rate that evenly divides
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48 kHz.
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Therefore, the value in the granule position field always counts samples
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assuming a 48 kHz decoding rate, and the rest of this specification makes
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the same assumption.
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</t>
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<t>
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The duration of an Opus packet as defined in <xref target="RFC6716"/> can be
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any multiple of 2.5 ms, up to a maximum of 120 ms.
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This duration is encoded in the TOC sequence at the beginning of each packet.
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The number of samples returned by a decoder corresponds to this duration
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exactly, even for the first few packets.
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For example, a 20 ms packet fed to a decoder running at 48 kHz will
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always return 960 samples.
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A demuxer can parse the TOC sequence at the beginning of each Ogg packet to
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work backwards or forwards from a packet with a known granule position (i.e.,
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the last packet completed on some page) in order to assign granule positions
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to every packet, or even every individual sample.
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The one exception is the last page in the stream, as described below.
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</t>
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<t>
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All other pages with completed packets after the first MUST have a granule
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position equal to the number of samples contained in packets that complete on
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that page plus the granule position of the most recent page with completed
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packets.
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This guarantees that a demuxer can assign individual packets the same granule
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position when working forwards as when working backwards.
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For this to work, there cannot be any gaps.
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</t>
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<section anchor="gap-repair" title="Repairing Gaps in Real-time Streams">
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<t>
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In order to support capturing a real-time stream that has lost or not
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transmitted packets, a multiplexer (muxer) SHOULD emit packets that explicitly
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request the use of Packet Loss Concealment (PLC) in place of the missing
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packets.
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Implementations that fail to do so still MUST NOT increment the granule
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position for a page by anything other than the number of samples contained in
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packets that actually complete on that page.
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</t>
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<t>
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Only gaps that are a multiple of 2.5 ms are repairable, as these are the
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only durations that can be created by packet loss or discontinuous
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transmission.
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Muxers need not handle other gap sizes.
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Creating the necessary packets involves synthesizing a TOC byte (defined in
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Section 3.1 of <xref target="RFC6716"/>)—and whatever
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additional internal framing is needed—to indicate the packet duration
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for each stream.
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The actual length of each missing Opus frame inside the packet is zero bytes,
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as defined in Section 3.2.1 of <xref target="RFC6716"/>.
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</t>
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<t>
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Zero-byte frames MAY be packed into packets using any of codes 0, 1,
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2, or 3.
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|
When successive frames have the same configuration, the higher code packings
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reduce overhead.
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|
Likewise, if the TOC configuration matches, the muxer MAY further combine the
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empty frames with previous or subsequent non-zero-length frames (using
|
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code 2 or VBR code 3).
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|
</t>
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<t>
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<xref target="RFC6716"/> does not impose any requirements on the PLC, but this
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|
section outlines choices that are expected to have a positive influence on
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most PLC implementations, including the reference implementation.
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Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth,
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channel count, and frame size as the previous packet (if any).
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|
This is the simplest and usually the most well-tested case for the PLC to
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handle and it covers all losses that do not include a configuration switch,
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|
as defined in Section 4.5 of <xref target="RFC6716"/>.
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|
</t>
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|
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<t>
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|
When a previous packet is available, keeping the audio bandwidth and channel
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|
count the same allows the PLC to provide maximum continuity in the concealment
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|
data it generates.
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|
However, if the size of the gap is not a multiple of the most recent frame
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size, then the frame size will have to change for at least some frames.
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|
Such changes SHOULD be delayed as long as possible to simplify
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|
things for PLC implementations.
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|
</t>
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<t>
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|
As an example, a 95 ms gap could be encoded as nineteen 5 ms frames
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|
in two bytes with a single CBR code 3 packet.
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|
If the previous frame size was 20 ms, using four 20 ms frames
|
|
followed by three 5 ms frames requires 4 bytes (plus an extra byte
|
|
of Ogg lacing overhead), but allows the PLC to use its well-tested steady
|
|
state behavior for as long as possible.
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|
The total bitrate of the latter approach, including Ogg overhead, is about
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|
0.4 kbps, so the impact on file size is minimal.
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|
</t>
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<t>
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Changing modes is discouraged, since this causes some decoder implementations
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to reset their PLC state.
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However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple
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of 10 ms.
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If switching to CELT mode is needed to match the gap size, a muxer SHOULD do
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so at the end of the gap to allow the PLC to function for as long as possible.
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</t>
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<t>
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In the example above, if the previous frame was a 20 ms SILK mode frame,
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|
the better solution is to synthesize a packet describing four 20 ms SILK
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|
frames, followed by a packet with a single 10 ms SILK
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|
frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms
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gap.
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|
This also requires four bytes to describe the synthesized packet data (two
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|
bytes for a CBR code 3 and one byte each for two code 0 packets) but three
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bytes of Ogg lacing overhead are needed to mark the packet boundaries.
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|
At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality
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solution.
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</t>
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<t>
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|
Since medium-band audio is an option only in the SILK mode, wideband frames
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|
SHOULD be generated if switching from that configuration to CELT mode, to
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|
ensure that any PLC implementation which does try to migrate state between
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|
the modes will be able to preserve all of the available audio bandwidth.
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|
</t>
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</section>
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|
|
<section anchor="preskip" title="Pre-skip">
|
|
<t>
|
|
There is some amount of latency introduced during the decoding process, to
|
|
allow for overlap in the CELT mode, stereo mixing in the SILK mode, and
|
|
resampling.
|
|
The encoder might have introduced additional latency through its own resampling
|
|
and analysis (though the exact amount is not specified).
|
|
Therefore, the first few samples produced by the decoder do not correspond to
|
|
real input audio, but are instead composed of padding inserted by the encoder
|
|
to compensate for this latency.
|
|
These samples need to be stored and decoded, as Opus is an asymptotically
|
|
convergent predictive codec, meaning the decoded contents of each frame depend
|
|
on the recent history of decoder inputs.
|
|
However, a player will want to skip these samples after decoding them.
|
|
</t>
|
|
|
|
<t>
|
|
A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals
|
|
the number of samples that SHOULD be skipped (decoded but discarded) at the
|
|
beginning of the stream, though some specific applications might have a reason
|
|
for looking at that data.
|
|
This amount need not be a multiple of 2.5 ms, MAY be smaller than a single
|
|
packet, or MAY span the contents of several packets.
|
|
These samples are not valid audio.
|
|
</t>
|
|
|
|
<t>
|
|
For example, if the first Opus frame uses the CELT mode, it will always
|
|
produce 120 samples of windowed overlap-add data.
|
|
However, the overlap data is initially all zeros (since there is no prior
|
|
frame), meaning this cannot, in general, accurately represent the original
|
|
audio.
|
|
The SILK mode requires additional delay to account for its analysis and
|
|
resampling latency.
|
|
The encoder delays the original audio to avoid this problem.
|
|
</t>
|
|
|
|
<t>
|
|
The pre-skip field MAY also be used to perform sample-accurate cropping of
|
|
already encoded streams.
|
|
In this case, a value of at least 3840 samples (80 ms) provides
|
|
sufficient history to the decoder that it will have converged
|
|
before the stream's output begins.
|
|
</t>
|
|
|
|
</section>
|
|
|
|
<section anchor="pcm_sample_position" title="PCM Sample Position">
|
|
<t>
|
|
The PCM sample position is determined from the granule position using the
|
|
formula
|
|
</t>
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
'PCM sample position' = 'granule position' - 'pre-skip' .
|
|
]]></artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
For example, if the granule position of the first audio data page is 59,971,
|
|
and the pre-skip is 11,971, then the PCM sample position of the last decoded
|
|
sample from that page is 48,000.
|
|
</t>
|
|
<t>
|
|
This can be converted into a playback time using the formula
|
|
</t>
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
'PCM sample position'
|
|
'playback time' = --------------------- .
|
|
48000.0
|
|
]]></artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
The initial PCM sample position before any samples are played is normally '0'.
|
|
In this case, the PCM sample position of the first audio sample to be played
|
|
starts at '1', because it marks the time on the clock
|
|
<spanx style="emph">after</spanx> that sample has been played, and a stream
|
|
that is exactly one second long has a final PCM sample position of '48000',
|
|
as in the example here.
|
|
</t>
|
|
|
|
<t>
|
|
Vorbis streams use a granule position smaller than the number of audio samples
|
|
contained in the first audio data page to indicate that some of those samples
|
|
are trimmed from the output (see <xref target="vorbis-trim"/>).
|
|
However, to do so, Vorbis requires that the first audio data page contains
|
|
exactly two packets, in order to allow the decoder to perform PCM position
|
|
adjustments before needing to return any PCM data.
|
|
Opus uses the pre-skip mechanism for this purpose instead, since the encoder
|
|
might introduce more than a single packet's worth of latency, and since very
|
|
large packets in streams with a very large number of channels might not fit
|
|
on a single page.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="end_trimming" title="End Trimming">
|
|
<t>
|
|
The page with the 'end of stream' flag set MAY have a granule position that
|
|
indicates the page contains less audio data than would normally be returned by
|
|
decoding up through the final packet.
|
|
This is used to end the stream somewhere other than an even frame boundary.
|
|
The granule position of the most recent audio data page with completed packets
|
|
is used to make this determination, or '0' is used if there were no previous
|
|
audio data pages with a completed packet.
|
|
The difference between these granule positions indicates how many samples to
|
|
keep after decoding the packets that completed on the final page.
|
|
The remaining samples are discarded.
|
|
The number of discarded samples SHOULD be no larger than the number decoded
|
|
from the last packet.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="start_granpos_restrictions"
|
|
title="Restrictions on the Initial Granule Position">
|
|
<t>
|
|
The granule position of the first audio data page with a completed packet MAY
|
|
be larger than the number of samples contained in packets that complete on
|
|
that page, however it MUST NOT be smaller, unless that page has the 'end of
|
|
stream' flag set.
|
|
Allowing a granule position larger than the number of samples allows the
|
|
beginning of a stream to be cropped or a live stream to be joined without
|
|
rewriting the granule position of all the remaining pages.
|
|
This means that the PCM sample position just before the first sample to be
|
|
played MAY be larger than '0'.
|
|
Synchronization when multiplexing with other logical streams still uses the PCM
|
|
sample position relative to '0' to compute sample times.
|
|
This does not affect the behavior of pre-skip: exactly 'pre-skip' samples
|
|
SHOULD be skipped from the beginning of the decoded output, even if the
|
|
initial PCM sample position is greater than zero.
|
|
</t>
|
|
|
|
<t>
|
|
On the other hand, a granule position that is smaller than the number of
|
|
decoded samples prevents a demuxer from working backwards to assign each
|
|
packet or each individual sample a valid granule position, since granule
|
|
positions are non-negative.
|
|
An implementation MUST treat any stream as invalid if the granule position
|
|
is smaller than the number of samples contained in packets that complete on
|
|
the first audio data page with a completed packet, unless that page has the
|
|
'end of stream' flag set.
|
|
It MAY defer this action until it decodes the last packet completed on that
|
|
page.
|
|
</t>
|
|
|
|
<t>
|
|
If that page has the 'end of stream' flag set, a demuxer MUST treat any stream
|
|
as invalid if its granule position is smaller than the 'pre-skip' amount.
|
|
This would indicate that there are more samples to be skipped from the initial
|
|
decoded output than exist in the stream.
|
|
If the granule position is smaller than the number of decoded samples produced
|
|
by the packets that complete on that page, then a demuxer MUST use an initial
|
|
granule position of '0', and can work forwards from '0' to timestamp
|
|
individual packets.
|
|
If the granule position is larger than the number of decoded samples available,
|
|
then the demuxer MUST still work backwards as described above, even if the
|
|
'end of stream' flag is set, to determine the initial granule position, and
|
|
thus the initial PCM sample position.
|
|
Both of these will be greater than '0' in this case.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="seeking_and_preroll" title="Seeking and Pre-roll">
|
|
<t>
|
|
Seeking in Ogg files is best performed using a bisection search for a page
|
|
whose granule position corresponds to a PCM position at or before the seek
|
|
target.
|
|
With appropriately weighted bisection, accurate seeking can be performed in
|
|
just one or two bisections on average, even in multi-gigabyte files.
|
|
See <xref target="seeking"/> for an example of general implementation guidance.
|
|
</t>
|
|
|
|
<t>
|
|
When seeking within an Ogg Opus stream, an implementation SHOULD start decoding
|
|
(and discarding the output) at least 3840 samples (80 ms) prior to
|
|
the seek target in order to ensure that the output audio is correct by the
|
|
time it reaches the seek target.
|
|
This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the
|
|
beginning of the stream.
|
|
If the point 80 ms prior to the seek target comes before the initial PCM
|
|
sample position, an implementation SHOULD start decoding from the beginning of
|
|
the stream, applying pre-skip as normal, regardless of whether the pre-skip is
|
|
larger or smaller than 80 ms, and then continue to discard samples
|
|
to reach the seek target (if any).
|
|
</t>
|
|
</section>
|
|
|
|
</section>
|
|
|
|
<section anchor="headers" title="Header Packets">
|
|
<t>
|
|
An Ogg Opus logical stream contains exactly two mandatory header packets:
|
|
an identification header and a comment header.
|
|
</t>
|
|
|
|
<section anchor="id_header" title="Identification Header">
|
|
|
|
<figure anchor="id_header_packet" title="ID Header Packet" align="center">
|
|
<artwork align="center"><![CDATA[
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| 'O' | 'p' | 'u' | 's' |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| 'H' | 'e' | 'a' | 'd' |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| Version = 1 | Channel Count | Pre-skip |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| Input Sample Rate (Hz) |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| Output Gain (Q7.8 in dB) | Mapping Family| |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ :
|
|
| |
|
|
: Optional Channel Mapping Table... :
|
|
| |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
]]></artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
The fields in the identification (ID) header have the following meaning:
|
|
<list style="numbers">
|
|
<t>Magic Signature:
|
|
<vspace blankLines="1"/>
|
|
This is an 8-octet (64-bit) field that allows codec identification and is
|
|
human-readable.
|
|
It contains, in order, the magic numbers:
|
|
<list style="empty">
|
|
<t>0x4F 'O'</t>
|
|
<t>0x70 'p'</t>
|
|
<t>0x75 'u'</t>
|
|
<t>0x73 's'</t>
|
|
<t>0x48 'H'</t>
|
|
<t>0x65 'e'</t>
|
|
<t>0x61 'a'</t>
|
|
<t>0x64 'd'</t>
|
|
</list>
|
|
Starting with "Op" helps distinguish it from audio data packets, as this is an
|
|
invalid TOC sequence.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Version (8 bits, unsigned):
|
|
<vspace blankLines="1"/>
|
|
The version number MUST always be '1' for this version of the encapsulation
|
|
specification.
|
|
Implementations SHOULD treat streams where the upper four bits of the version
|
|
number match that of a recognized specification as backwards-compatible with
|
|
that specification.
|
|
That is, the version number can be split into "major" and "minor" version
|
|
sub-fields, with changes to the "minor" sub-field (in the lower four bits)
|
|
signaling compatible changes.
|
|
For example, an implementation of this specification SHOULD accept any stream
|
|
with a version number of '15' or less, and SHOULD assume any stream with a
|
|
version number '16' or greater is incompatible.
|
|
The initial version '1' was chosen to keep implementations from relying on this
|
|
octet as a null terminator for the "OpusHead" string.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Output Channel Count 'C' (8 bits, unsigned):
|
|
<vspace blankLines="1"/>
|
|
This is the number of output channels.
|
|
This might be different than the number of encoded channels, which can change
|
|
on a packet-by-packet basis.
|
|
This value MUST NOT be zero.
|
|
The maximum allowable value depends on the channel mapping family, and might be
|
|
as large as 255.
|
|
See <xref target="channel_mapping"/> for details.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Pre-skip (16 bits, unsigned, little
|
|
endian):
|
|
<vspace blankLines="1"/>
|
|
This is the number of samples (at 48 kHz) to discard from the decoder
|
|
output when starting playback, and also the number to subtract from a page's
|
|
granule position to calculate its PCM sample position.
|
|
When cropping the beginning of existing Ogg Opus streams, a pre-skip of at
|
|
least 3,840 samples (80 ms) is RECOMMENDED to ensure complete
|
|
convergence in the decoder.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Input Sample Rate (32 bits, unsigned, little
|
|
endian):
|
|
<vspace blankLines="1"/>
|
|
This is the sample rate of the original input (before encoding), in Hz.
|
|
This field is <spanx style="emph">not</spanx> the sample rate to use for
|
|
playback of the encoded data.
|
|
<vspace blankLines="1"/>
|
|
Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and
|
|
20 kHz.
|
|
Each packet in the stream can have a different audio bandwidth.
|
|
Regardless of the audio bandwidth, the reference decoder supports decoding any
|
|
stream at a sample rate of 8, 12, 16, 24, or 48 kHz.
|
|
The original sample rate of the audio passed to the encoder is not preserved
|
|
by the lossy compression.
|
|
<vspace blankLines="1"/>
|
|
An Ogg Opus player SHOULD select the playback sample rate according to the
|
|
following procedure:
|
|
<list style="numbers">
|
|
<t>If the hardware supports 48 kHz playback, decode at 48 kHz.</t>
|
|
<t>Otherwise, if the hardware's highest available sample rate is a supported
|
|
rate, decode at this sample rate.</t>
|
|
<t>Otherwise, if the hardware's highest available sample rate is less than
|
|
48 kHz, decode at the next higher Opus supported rate above the highest
|
|
available hardware rate and resample.</t>
|
|
<t>Otherwise, decode at 48 kHz and resample.</t>
|
|
</list>
|
|
However, the 'Input Sample Rate' field allows the muxer to pass the sample
|
|
rate of the original input stream as metadata.
|
|
This is useful when the user requires the output sample rate to match the
|
|
input sample rate.
|
|
For example, when not playing the output, an implementation writing PCM format
|
|
samples to disk might choose to resample the audio back to the original input
|
|
sample rate to reduce surprise to the user, who might reasonably expect to get
|
|
back a file with the same sample rate.
|
|
<vspace blankLines="1"/>
|
|
A value of zero indicates 'unspecified'.
|
|
Muxers SHOULD write the actual input sample rate or zero, but implementations
|
|
which do something with this field SHOULD take care to behave sanely if given
|
|
crazy values (e.g., do not actually upsample the output to 10 MHz if
|
|
requested).
|
|
Implementations SHOULD support input sample rates between 8 kHz and
|
|
192 kHz (inclusive).
|
|
Rates outside this range MAY be ignored by falling back to the default rate of
|
|
48 kHz instead.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Output Gain (16 bits, signed, little endian):
|
|
<vspace blankLines="1"/>
|
|
This is a gain to be applied when decoding.
|
|
It is 20*log10 of the factor by which to scale the decoder output to achieve
|
|
the desired playback volume, stored in a 16-bit, signed, two's complement
|
|
fixed-point value with 8 fractional bits (i.e.,
|
|
Q7.8 <xref target="q-notation"/>).
|
|
<vspace blankLines="1"/>
|
|
To apply the gain, an implementation could use
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
sample *= pow(10, output_gain/(20.0*256)) ,
|
|
]]></artwork>
|
|
</figure>
|
|
where output_gain is the raw 16-bit value from the header.
|
|
<vspace blankLines="1"/>
|
|
Players and media frameworks SHOULD apply it by default.
|
|
If a player chooses to apply any volume adjustment or gain modification, such
|
|
as the R128_TRACK_GAIN (see <xref target="comment_header"/>), the adjustment
|
|
MUST be applied in addition to this output gain in order to achieve playback
|
|
at the normalized volume.
|
|
<vspace blankLines="1"/>
|
|
A muxer SHOULD set this field to zero, and instead apply any gain prior to
|
|
encoding, when this is possible and does not conflict with the user's wishes.
|
|
A nonzero output gain indicates the gain was adjusted after encoding, or that
|
|
a user wished to adjust the gain for playback while preserving the ability
|
|
to recover the original signal amplitude.
|
|
<vspace blankLines="1"/>
|
|
Although the output gain has enormous range (+/- 128 dB, enough to amplify
|
|
inaudible sounds to the threshold of physical pain), most applications can
|
|
only reasonably use a small portion of this range around zero.
|
|
The large range serves in part to ensure that gain can always be losslessly
|
|
transferred between OpusHead and R128 gain tags (see below) without
|
|
saturating.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Channel Mapping Family (8 bits, unsigned):
|
|
<vspace blankLines="1"/>
|
|
This octet indicates the order and semantic meaning of the output channels.
|
|
<vspace blankLines="1"/>
|
|
Each currently specified value of this octet indicates a mapping family, which
|
|
defines a set of allowed channel counts, and the ordered set of channel names
|
|
for each allowed channel count.
|
|
The details are described in <xref target="channel_mapping"/>.
|
|
</t>
|
|
<t>Channel Mapping Table:
|
|
This table defines the mapping from encoded streams to output channels.
|
|
Its contents are specified in <xref target="channel_mapping"/>.
|
|
</t>
|
|
</list>
|
|
</t>
|
|
|
|
<t>
|
|
All fields in the ID headers are REQUIRED, except for the channel mapping
|
|
table, which MUST be omitted when the channel mapping family is 0, but
|
|
is REQUIRED otherwise.
|
|
Implementations SHOULD treat a stream as invalid if it contains an ID header
|
|
that does not have enough data for these fields, even if it contain a valid
|
|
Magic Signature.
|
|
Future versions of this specification, even backwards-compatible versions,
|
|
might include additional fields in the ID header.
|
|
If an ID header has a compatible major version, but a larger minor version,
|
|
an implementation MUST NOT treat it as invalid for containing additional data
|
|
not specified here, provided it still completes on the first page.
|
|
</t>
|
|
|
|
<section anchor="channel_mapping" title="Channel Mapping">
|
|
<t>
|
|
An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly
|
|
larger number of decoded channels (M + N) to yet another number of
|
|
output channels (C), which might be larger or smaller than the number of
|
|
decoded channels.
|
|
The order and meaning of these channels are defined by a channel mapping,
|
|
which consists of the 'channel mapping family' octet and, for channel mapping
|
|
families other than family 0, a channel mapping table, as illustrated in
|
|
<xref target="channel_mapping_table"/>.
|
|
</t>
|
|
|
|
<figure anchor="channel_mapping_table" title="Channel Mapping Table"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+
|
|
| Stream Count |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| Coupled Count | Channel Mapping... :
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
]]></artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
The fields in the channel mapping table have the following meaning:
|
|
<list style="numbers" counter="8">
|
|
<t>Stream Count 'N' (8 bits, unsigned):
|
|
<vspace blankLines="1"/>
|
|
This is the total number of streams encoded in each Ogg packet.
|
|
This value is necessary to correctly parse the packed Opus packets inside an
|
|
Ogg packet, as described in <xref target="packet_organization"/>.
|
|
This value MUST NOT be zero, as without at least one Opus packet with a valid
|
|
TOC sequence, a demuxer cannot recover the duration of an Ogg packet.
|
|
<vspace blankLines="1"/>
|
|
For channel mapping family 0, this value defaults to 1, and is not coded.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Coupled Stream Count 'M' (8 bits, unsigned):
|
|
This is the number of streams whose decoders are to be configured to produce
|
|
two channels (stereo).
|
|
This MUST be no larger than the total number of streams, N.
|
|
<vspace blankLines="1"/>
|
|
Each packet in an Opus stream has an internal channel count of 1 or 2, which
|
|
can change from packet to packet.
|
|
This is selected by the encoder depending on the bitrate and the audio being
|
|
encoded.
|
|
The original channel count of the audio passed to the encoder is not
|
|
necessarily preserved by the lossy compression.
|
|
<vspace blankLines="1"/>
|
|
Regardless of the internal channel count, any Opus stream can be decoded as
|
|
mono (a single channel) or stereo (two channels) by appropriate initialization
|
|
of the decoder.
|
|
The 'coupled stream count' field indicates that the decoders for the first M
|
|
Opus streams are to be initialized for stereo (two-channel) output, and the
|
|
remaining (N - M) decoders are to be initialized for mono (a single
|
|
channel) only.
|
|
The total number of decoded channels, (M + N), MUST be no larger than
|
|
255, as there is no way to index more channels than that in the channel
|
|
mapping.
|
|
<vspace blankLines="1"/>
|
|
For channel mapping family 0, this value defaults to (C - 1)
|
|
(i.e., 0 for mono and 1 for stereo), and is not coded.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Channel Mapping (8*C bits):
|
|
This contains one octet per output channel, indicating which decoded channel
|
|
is to be used for each one.
|
|
Let 'index' be the value of this octet for a particular output channel.
|
|
This value MUST either be smaller than (M + N), or be the special
|
|
value 255.
|
|
If 'index' is less than 2*M, the output MUST be taken from decoding stream
|
|
('index'/2) as stereo and selecting the left channel if 'index' is even, and
|
|
the right channel if 'index' is odd.
|
|
If 'index' is 2*M or larger, but less than 255, the output MUST be taken from
|
|
decoding stream ('index' - M) as mono.
|
|
If 'index' is 255, the corresponding output channel MUST contain pure silence.
|
|
<vspace blankLines="1"/>
|
|
The number of output channels, C, is not constrained to match the number of
|
|
decoded channels (M + N).
|
|
A single index value MAY appear multiple times, i.e., the same decoded channel
|
|
might be mapped to multiple output channels.
|
|
Some decoded channels might not be assigned to any output channel, as well.
|
|
<vspace blankLines="1"/>
|
|
For channel mapping family 0, the first index defaults to 0, and if
|
|
C == 2, the second index defaults to 1.
|
|
Neither index is coded.
|
|
</t>
|
|
</list>
|
|
</t>
|
|
|
|
<t>
|
|
After producing the output channels, the channel mapping family determines the
|
|
semantic meaning of each one.
|
|
There are three defined mapping families in this specification.
|
|
</t>
|
|
|
|
<section anchor="channel_mapping_0" title="Channel Mapping Family 0">
|
|
<t>
|
|
Allowed numbers of channels: 1 or 2.
|
|
RTP mapping.
|
|
This is the same channel interpretation as <xref target="RFC7587"/>.
|
|
</t>
|
|
<t>
|
|
<list style="symbols">
|
|
<t>1 channel: monophonic (mono).</t>
|
|
<t>2 channels: stereo (left, right).</t>
|
|
</list>
|
|
Special mapping: This channel mapping value also
|
|
indicates that the contents consists of a single Opus stream that is stereo if
|
|
and only if C == 2, with stream index 0 mapped to output
|
|
channel 0 (mono, or left channel) and stream index 1 mapped to
|
|
output channel 1 (right channel) if stereo.
|
|
When the 'channel mapping family' octet has this value, the channel mapping
|
|
table MUST be omitted from the ID header packet.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="channel_mapping_1" title="Channel Mapping Family 1">
|
|
<t>
|
|
Allowed numbers of channels: 1...8.
|
|
Vorbis channel order (see below).
|
|
</t>
|
|
<t>
|
|
Each channel is assigned to a speaker location in a conventional surround
|
|
arrangement.
|
|
Specific locations depend on the number of channels, and are given below
|
|
in order of the corresponding channel indices.
|
|
<list style="symbols">
|
|
<t>1 channel: monophonic (mono).</t>
|
|
<t>2 channels: stereo (left, right).</t>
|
|
<t>3 channels: linear surround (left, center, right)</t>
|
|
<t>4 channels: quadraphonic (front left, front right, rear left, rear right).</t>
|
|
<t>5 channels: 5.0 surround (front left, front center, front right, rear left, rear right).</t>
|
|
<t>6 channels: 5.1 surround (front left, front center, front right, rear left, rear right, LFE).</t>
|
|
<t>7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE).</t>
|
|
<t>8 channels: 7.1 surround (front left, front center, front right, side left, side right, rear left, rear right, LFE)</t>
|
|
</list>
|
|
</t>
|
|
<t>
|
|
This set of surround options and speaker location orderings is the same
|
|
as those used by the Vorbis codec <xref target="vorbis-mapping"/>.
|
|
The ordering is different from the one used by the
|
|
WAVE <xref target="wave-multichannel"/> and
|
|
Free Lossless Audio Codec (FLAC) <xref target="flac"/> formats,
|
|
so correct ordering requires permutation of the output channels when decoding
|
|
to or encoding from those formats.
|
|
'LFE' here refers to a Low Frequency Effects channel, often mapped to a
|
|
subwoofer with no particular spatial position.
|
|
Implementations SHOULD identify 'side' or 'rear' speaker locations with
|
|
'surround' and 'back' as appropriate when interfacing with audio formats
|
|
or systems which prefer that terminology.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="channel_mapping_255"
|
|
title="Channel Mapping Family 255">
|
|
<t>
|
|
Allowed numbers of channels: 1...255.
|
|
No defined channel meaning.
|
|
</t>
|
|
<t>
|
|
Channels are unidentified.
|
|
General-purpose players SHOULD NOT attempt to play these streams.
|
|
Offline implementations MAY deinterleave the output into separate PCM files,
|
|
one per channel.
|
|
Implementations SHOULD NOT produce output for channels mapped to stream index
|
|
255 (pure silence) unless they have no other way to indicate the index of
|
|
non-silent channels.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="channel_mapping_undefined"
|
|
title="Undefined Channel Mappings">
|
|
<t>
|
|
The remaining channel mapping families (2...254) are reserved.
|
|
A demuxer implementation encountering a reserved channel mapping family value
|
|
SHOULD act as though the value is 255.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="downmix" title="Downmixing">
|
|
<t>
|
|
An Ogg Opus player MUST support any valid channel mapping with a channel
|
|
mapping family of 0 or 1, even if the number of channels does not match the
|
|
physically connected audio hardware.
|
|
Players SHOULD perform channel mixing to increase or reduce the number of
|
|
channels as needed.
|
|
</t>
|
|
|
|
<t>
|
|
Implementations MAY use the matrices in
|
|
Figures <xref target="downmix-matrix-3" format="counter"/>
|
|
through <xref target="downmix-matrix-8" format="counter"/> to implement
|
|
downmixing from multichannel files using
|
|
<xref target="channel_mapping_1">Channel Mapping Family 1</xref>, which are
|
|
known to give acceptable results for stereo.
|
|
Matrices for 3 and 4 channels are normalized so each coefficient row sums
|
|
to 1 to avoid clipping.
|
|
For 5 or more channels they are normalized to 2 as a compromise between
|
|
clipping and dynamic range reduction.
|
|
</t>
|
|
<t>
|
|
In these matrices the front left and front right channels are generally
|
|
passed through directly.
|
|
When a surround channel is split between both the left and right stereo
|
|
channels, coefficients are chosen so their squares sum to 1, which
|
|
helps preserve the perceived intensity.
|
|
Rear channels are mixed more diffusely or attenuated to maintain focus
|
|
on the front channels.
|
|
</t>
|
|
|
|
<figure anchor="downmix-matrix-3"
|
|
title="Stereo downmix matrix for the linear surround channel mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
L output = ( 0.585786 * left + 0.414214 * center )
|
|
R output = ( 0.414214 * center + 0.585786 * right )
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1 and 1/sqrt(2), multiplied by
|
|
1/(1 + 1/sqrt(2)) for normalization.
|
|
</postamble>
|
|
</figure>
|
|
|
|
<figure anchor="downmix-matrix-4"
|
|
title="Stereo downmix matrix for the quadraphonic channel mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
/ \ / \ / FL \
|
|
| L output | | 0.422650 0.000000 0.366025 0.211325 | | FR |
|
|
| R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL |
|
|
\ / \ / \ RR /
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
|
|
1/(1 + sqrt(3)/2 + 1/2) for normalization.
|
|
</postamble>
|
|
</figure>
|
|
|
|
<figure anchor="downmix-matrix-5"
|
|
title="Stereo downmix matrix for the 5.0 surround mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
/ FL \
|
|
/ \ / \ | FC |
|
|
| L | | 0.650802 0.460186 0.000000 0.563611 0.325401 | | FR |
|
|
| R | = | 0.000000 0.460186 0.650802 0.325401 0.563611 | | RL |
|
|
\ / \ / | RR |
|
|
\ /
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
|
|
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2)
|
|
for normalization.
|
|
</postamble>
|
|
</figure>
|
|
|
|
<figure anchor="downmix-matrix-6"
|
|
title="Stereo downmix matrix for the 5.1 surround mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
/FL \
|
|
/ \ / \ |FC |
|
|
|L| | 0.529067 0.374107 0.000000 0.458186 0.264534 0.374107 | |FR |
|
|
|R| = | 0.000000 0.374107 0.529067 0.264534 0.458186 0.374107 | |RL |
|
|
\ / \ / |RR |
|
|
\LFE/
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
|
|
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2))
|
|
for normalization.
|
|
</postamble>
|
|
</figure>
|
|
|
|
<figure anchor="downmix-matrix-7"
|
|
title="Stereo downmix matrix for the 6.1 surround mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
/ \
|
|
| 0.455310 0.321953 0.000000 0.394310 0.227655 0.278819 0.321953 |
|
|
| 0.000000 0.321953 0.455310 0.227655 0.394310 0.278819 0.321953 |
|
|
\ /
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and
|
|
sqrt(3)/2/sqrt(2), multiplied by
|
|
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 +
|
|
sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization.
|
|
The coefficients are in the same order as in <xref target="channel_mapping_1" />,
|
|
and the matrices above.
|
|
</postamble>
|
|
</figure>
|
|
|
|
<figure anchor="downmix-matrix-8"
|
|
title="Stereo downmix matrix for the 7.1 surround mapping"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
/ \
|
|
| .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 |
|
|
| .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 |
|
|
\ /
|
|
]]></artwork>
|
|
<postamble>
|
|
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
|
|
2/(2 + 2/sqrt(2) + sqrt(3)) for normalization.
|
|
The coefficients are in the same order as in <xref target="channel_mapping_1" />,
|
|
and the matrices above.
|
|
</postamble>
|
|
</figure>
|
|
|
|
</section>
|
|
|
|
</section> <!-- end channel_mapping_table -->
|
|
|
|
</section> <!-- end id_header -->
|
|
|
|
<section anchor="comment_header" title="Comment Header">
|
|
|
|
<figure anchor="comment_header_packet" title="Comment Header Packet"
|
|
align="center">
|
|
<artwork align="center"><![CDATA[
|
|
0 1 2 3
|
|
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| 'O' | 'p' | 'u' | 's' |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| 'T' | 'a' | 'g' | 's' |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| Vendor String Length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| |
|
|
: Vendor String... :
|
|
| |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| User Comment List Length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| User Comment #0 String Length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| |
|
|
: User Comment #0 String... :
|
|
| |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
| User Comment #1 String Length |
|
|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
: :
|
|
]]></artwork>
|
|
</figure>
|
|
|
|
<t>
|
|
The comment header consists of a 64-bit magic signature, followed by data in
|
|
the same format as the <xref target="vorbis-comment"/> header used in Ogg
|
|
Vorbis, except (like Ogg Theora and Speex) the final "framing bit" specified
|
|
in the Vorbis spec is not present.
|
|
<list style="numbers">
|
|
<t>Magic Signature:
|
|
<vspace blankLines="1"/>
|
|
This is an 8-octet (64-bit) field that allows codec identification and is
|
|
human-readable.
|
|
It contains, in order, the magic numbers:
|
|
<list style="empty">
|
|
<t>0x4F 'O'</t>
|
|
<t>0x70 'p'</t>
|
|
<t>0x75 'u'</t>
|
|
<t>0x73 's'</t>
|
|
<t>0x54 'T'</t>
|
|
<t>0x61 'a'</t>
|
|
<t>0x67 'g'</t>
|
|
<t>0x73 's'</t>
|
|
</list>
|
|
Starting with "Op" helps distinguish it from audio data packets, as this is an
|
|
invalid TOC sequence.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Vendor String Length (32 bits, unsigned, little endian):
|
|
<vspace blankLines="1"/>
|
|
This field gives the length of the following vendor string, in octets.
|
|
It MUST NOT indicate that the vendor string is longer than the rest of the
|
|
packet.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>Vendor String (variable length, UTF-8 vector):
|
|
<vspace blankLines="1"/>
|
|
This is a simple human-readable tag for vendor information, encoded as a UTF-8
|
|
string <xref target="RFC3629"/>.
|
|
No terminating null octet is necessary.
|
|
<vspace blankLines="1"/>
|
|
This tag is intended to identify the codec encoder and encapsulation
|
|
implementations, for tracing differences in technical behavior.
|
|
User-facing applications can use the 'ENCODER' user comment tag to identify
|
|
themselves.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>User Comment List Length (32 bits, unsigned, little endian):
|
|
<vspace blankLines="1"/>
|
|
This field indicates the number of user-supplied comments.
|
|
It MAY indicate there are zero user-supplied comments, in which case there are
|
|
no additional fields in the packet.
|
|
It MUST NOT indicate that there are so many comments that the comment string
|
|
lengths would require more data than is available in the rest of the packet.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>User Comment #i String Length (32 bits, unsigned, little endian):
|
|
<vspace blankLines="1"/>
|
|
This field gives the length of the following user comment string, in octets.
|
|
There is one for each user comment indicated by the 'user comment list length'
|
|
field.
|
|
It MUST NOT indicate that the string is longer than the rest of the packet.
|
|
<vspace blankLines="1"/>
|
|
</t>
|
|
<t>User Comment #i String (variable length, UTF-8 vector):
|
|
<vspace blankLines="1"/>
|
|
This field contains a single user comment encoded as a UTF-8
|
|
string <xref target="RFC3629"/>.
|
|
There is one for each user comment indicated by the 'user comment list length'
|
|
field.
|
|
</t>
|
|
</list>
|
|
</t>
|
|
|
|
<t>
|
|
The vendor string length and user comment list length are REQUIRED, and
|
|
implementations SHOULD treat a stream as invalid if it contains a comment
|
|
header that does not have enough data for these fields, or that does not
|
|
contain enough data for the corresponding vendor string or user comments they
|
|
describe.
|
|
Making this check before allocating the associated memory to contain the data
|
|
helps prevent a possible Denial-of-Service (DoS) attack from small comment
|
|
headers that claim to contain strings longer than the entire packet or more
|
|
user comments than than could possibly fit in the packet.
|
|
</t>
|
|
|
|
<t>
|
|
Immediately following the user comment list, the comment header MAY
|
|
contain zero-padding or other binary data which is not specified here.
|
|
If the least-significant bit of the first byte of this data is 1, then editors
|
|
SHOULD preserve the contents of this data when updating the tags, but if this
|
|
bit is 0, all such data MAY be treated as padding, and truncated or discarded
|
|
as desired.
|
|
This allows informal experimentation with the format of this binary data until
|
|
it can be specified later.
|
|
</t>
|
|
|
|
<t>
|
|
The comment header can be arbitrarily large and might be spread over a large
|
|
number of Ogg pages.
|
|
Implementations MUST avoid attempting to allocate excessive amounts of memory
|
|
when presented with a very large comment header.
|
|
To accomplish this, implementations MAY treat a stream as invalid if it has a
|
|
comment header larger than 125,829,120 octets (120 MB), and MAY
|
|
ignore individual comments that are not fully contained within the first
|
|
61,440 octets of the comment header.
|
|
</t>
|
|
|
|
<section anchor="comment_format" title="Tag Definitions">
|
|
<t>
|
|
The user comment strings follow the NAME=value format described by
|
|
<xref target="vorbis-comment"/> with the same recommended tag names:
|
|
ARTIST, TITLE, DATE, ALBUM, and so on.
|
|
</t>
|
|
<t>
|
|
Two new comment tags are introduced here:
|
|
</t>
|
|
|
|
<t>First, an optional gain for track normalization:</t>
|
|
<figure align="center">
|
|
<artwork align="left"><![CDATA[
|
|
R128_TRACK_GAIN=-573
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
representing the volume shift needed to normalize the track's volume
|
|
during isolated playback, in random shuffle, and so on.
|
|
The gain is a Q7.8 fixed point number in dB, as in the ID header's 'output
|
|
gain' field.
|
|
This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in
|
|
Vorbis <xref target="replay-gain"/>, except that the normal volume
|
|
reference is the <xref target="EBU-R128"/> standard.
|
|
</t>
|
|
<t>Second, an optional gain for album normalization:</t>
|
|
<figure align="center">
|
|
<artwork align="left"><![CDATA[
|
|
R128_ALBUM_GAIN=111
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
representing the volume shift needed to normalize the overall volume when
|
|
played as part of a particular collection of tracks.
|
|
The gain is also a Q7.8 fixed point number in dB, as in the ID header's
|
|
'output gain' field.
|
|
The values '-573' and '111' given here are just examples.
|
|
</t>
|
|
<t>
|
|
An Ogg Opus stream MUST NOT have more than one of each of these tags, and if
|
|
present their values MUST be an integer from -32768 to 32767, inclusive,
|
|
represented in ASCII as a base 10 number with no whitespace.
|
|
A leading '+' or '-' character is valid.
|
|
Leading zeros are also permitted, but the value MUST be represented by
|
|
no more than 6 characters.
|
|
Other non-digit characters MUST NOT be present.
|
|
</t>
|
|
<t>
|
|
If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent
|
|
the R128 normalization gain relative to the 'output gain' field specified
|
|
in the ID header.
|
|
If a player chooses to make use of the R128_TRACK_GAIN tag or the
|
|
R128_ALBUM_GAIN tag, it MUST apply those gains
|
|
<spanx style="emph">in addition</spanx> to the 'output gain' value.
|
|
If a tool modifies the ID header's 'output gain' field, it MUST also update or
|
|
remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present.
|
|
A muxer SHOULD place the gain it wants other tools to use by default into the
|
|
'output gain' field, and not the comment tag.
|
|
</t>
|
|
<t>
|
|
To avoid confusion with multiple normalization schemes, an Opus comment header
|
|
SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK,
|
|
REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags, unless they are only
|
|
to be used in some context where there is guaranteed to be no such confusion.
|
|
<xref target="EBU-R128"/> normalization is preferred to the earlier
|
|
REPLAYGAIN schemes because of its clear definition and adoption by industry.
|
|
Peak normalizations are difficult to calculate reliably for lossy codecs
|
|
because of variation in excursion heights due to decoder differences.
|
|
In the authors' investigations they were not applied consistently or broadly
|
|
enough to merit inclusion here.
|
|
</t>
|
|
</section> <!-- end comment_format -->
|
|
</section> <!-- end comment_header -->
|
|
|
|
</section> <!-- end headers -->
|
|
|
|
<section anchor="packet_size_limits" title="Packet Size Limits">
|
|
<t>
|
|
Technically, valid Opus packets can be arbitrarily large due to the padding
|
|
format, although the amount of non-padding data they can contain is bounded.
|
|
These packets might be spread over a similarly enormous number of Ogg pages.
|
|
When encoding, implementations SHOULD limit the use of padding in audio data
|
|
packets to no more than is necessary to make a variable bitrate (VBR) stream
|
|
constant bitrate (CBR), unless they have no reasonable way to determine what
|
|
is necessary.
|
|
Demuxers SHOULD treat audio data packets as invalid (treat them as if they were
|
|
malformed Opus packets with an invalid TOC sequence) if they are larger than
|
|
61,440 octets per Opus stream, unless they have a specific reason for
|
|
allowing extra padding.
|
|
Such packets necessarily contain more padding than needed to make a stream CBR.
|
|
Demuxers MUST avoid attempting to allocate excessive amounts of memory when
|
|
presented with a very large packet.
|
|
Demuxers MAY treat audio data packets as invalid or partially process them if
|
|
they are larger than 61,440 octets in an Ogg Opus stream with channel
|
|
mapping families 0 or 1.
|
|
Demuxers MAY treat audio data packets as invalid or partially process them in
|
|
any Ogg Opus stream if the packet is larger than 61,440 octets and also
|
|
larger than 7,680 octets per Opus stream.
|
|
The presence of an extremely large packet in the stream could indicate a
|
|
memory exhaustion attack or stream corruption.
|
|
</t>
|
|
<t>
|
|
In an Ogg Opus stream, the largest possible valid packet that does not use
|
|
padding has a size of (61,298*N - 2) octets.
|
|
With 255 streams, this is 15,630,988 octets and can
|
|
span up to 61,298 Ogg pages, all but one of which will have a granule
|
|
position of -1.
|
|
This is of course a very extreme packet, consisting of 255 streams, each
|
|
containing 120 ms of audio encoded as 2.5 ms frames, each frame
|
|
using the maximum possible number of octets (1275) and stored in the least
|
|
efficient manner allowed (a VBR code 3 Opus packet).
|
|
Even in such a packet, most of the data will be zeros as 2.5 ms frames
|
|
cannot actually use all 1275 octets.
|
|
</t>
|
|
<t>
|
|
The largest packet consisting of entirely useful data is
|
|
(15,326*N - 2) octets.
|
|
This corresponds to 120 ms of audio encoded as 10 ms frames in either
|
|
SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little
|
|
sense for the quality achieved.
|
|
</t>
|
|
<t>
|
|
A more reasonable limit is (7,664*N - 2) octets.
|
|
This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode
|
|
frames, with a total bitrate just under 511 kbps (not counting the Ogg
|
|
encapsulation overhead).
|
|
For channel mapping family 1, N=8 provides a reasonable upper bound, as it
|
|
allows for each of the 8 possible output channels to be decoded from a
|
|
separate stereo Opus stream.
|
|
This gives a size of 61,310 octets, which is rounded up to a multiple of
|
|
1,024 octets to yield the audio data packet size of 61,440 octets
|
|
that any implementation is expected to be able to process successfully.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="encoder" title="Encoder Guidelines">
|
|
<t>
|
|
When encoding Opus streams, Ogg muxers SHOULD take into account the
|
|
algorithmic delay of the Opus encoder.
|
|
</t>
|
|
<t>
|
|
In encoders derived from the reference
|
|
implementation <xref target="RFC6716"/>, the number of samples can be
|
|
queried with:
|
|
</t>
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples));
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
To achieve good quality in the very first samples of a stream, implementations
|
|
MAY use linear predictive coding (LPC) extrapolation to generate at least 120
|
|
extra samples at the beginning to avoid the Opus encoder having to encode a
|
|
discontinuous signal.
|
|
For more information on linear prediction, see
|
|
<xref target="linear-prediction"/>.
|
|
For an input file containing 'length' samples, the implementation SHOULD set
|
|
the pre-skip header value to (delay_samples + extra_samples), encode
|
|
at least (length + delay_samples + extra_samples)
|
|
samples, and set the granule position of the last page to
|
|
(length + delay_samples + extra_samples).
|
|
This ensures that the encoded file has the same duration as the original, with
|
|
no time offset. The best way to pad the end of the stream is to also use LPC
|
|
extrapolation, but zero-padding is also acceptable.
|
|
</t>
|
|
|
|
<section anchor="lpc" title="LPC Extrapolation">
|
|
<t>
|
|
The first step in LPC extrapolation is to compute linear prediction
|
|
coefficients. <xref target="lpc-sample"/>
|
|
When extending the end of the signal, order-N (typically with N ranging from 8
|
|
to 40) LPC analysis is performed on a window near the end of the signal.
|
|
The last N samples are used as memory to an infinite impulse response (IIR)
|
|
filter.
|
|
</t>
|
|
<t>
|
|
The filter is then applied on a zero input to extrapolate the end of the signal.
|
|
Let a(k) be the kth LPC coefficient and x(n) be the nth sample of the signal,
|
|
each new sample past the end of the signal is computed as:
|
|
</t>
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
N
|
|
---
|
|
x(n) = \ a(k)*x(n-k)
|
|
/
|
|
---
|
|
k=1
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
The process is repeated independently for each channel.
|
|
It is possible to extend the beginning of the signal by applying the same
|
|
process backward in time.
|
|
When extending the beginning of the signal, it is best to apply a "fade in" to
|
|
the extrapolated signal, e.g. by multiplying it by a half-Hanning window
|
|
<xref target="hanning"/>.
|
|
</t>
|
|
|
|
</section>
|
|
|
|
<section anchor="continuous_chaining" title="Continuous Chaining">
|
|
<t>
|
|
In some applications, such as Internet radio, it is desirable to cut a long
|
|
stream into smaller chains, e.g. so the comment header can be updated.
|
|
This can be done simply by separating the input streams into segments and
|
|
encoding each segment independently.
|
|
The drawback of this approach is that it creates a small discontinuity
|
|
at the boundary due to the lossy nature of Opus.
|
|
A muxer MAY avoid this discontinuity by using the following procedure:
|
|
<list style="numbers">
|
|
<t>Encode the last frame of the first segment as an independent frame by
|
|
turning off all forms of inter-frame prediction.
|
|
De-emphasis is allowed.</t>
|
|
<t>Set the granule position of the last page to a point near the end of the
|
|
last frame.</t>
|
|
<t>Begin the second segment with a copy of the last frame of the first
|
|
segment.</t>
|
|
<t>Set the pre-skip value of the second stream in such a way as to properly
|
|
join the two streams.</t>
|
|
<t>Continue the encoding process normally from there, without any reset to
|
|
the encoder.</t>
|
|
</list>
|
|
</t>
|
|
<t>
|
|
In encoders derived from the reference implementation, inter-frame prediction
|
|
can be turned off by calling:
|
|
</t>
|
|
<figure align="center">
|
|
<artwork align="center"><![CDATA[
|
|
opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
For best results, this implementation requires that prediction be explicitly
|
|
enabled again before resuming normal encoding, even after a reset.
|
|
</t>
|
|
|
|
</section>
|
|
|
|
</section>
|
|
|
|
<section anchor="implementation" title="Implementation Status">
|
|
<t>
|
|
A brief summary of major implementations of this draft is available
|
|
at <eref target="https://wiki.xiph.org/OggOpusImplementation"/>,
|
|
along with their status.
|
|
</t>
|
|
<t>
|
|
[Note to RFC Editor: please remove this entire section before
|
|
final publication per <xref target="RFC6982"/>, along with
|
|
its references.]
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="security" title="Security Considerations">
|
|
<t>
|
|
Implementations of the Opus codec need to take appropriate security
|
|
considerations into account, as outlined in <xref target="RFC4732"/>.
|
|
This is just as much a problem for the container as it is for the codec itself.
|
|
Malicious payloads and/or input streams can be used to attack codec
|
|
implementations.
|
|
Implementations MUST NOT overrun their allocated memory nor consume excessive
|
|
resources when decoding payloads or processing input streams.
|
|
Although problems in encoding applications are typically rarer, this still
|
|
applies to a muxer, as vulnerabilities would allow an attacker to attack
|
|
transcoding gateways.
|
|
</t>
|
|
|
|
<t>
|
|
Header parsing code contains the most likely area for potential overruns.
|
|
It is important for implementations to ensure their buffers contain enough
|
|
data for all of the required fields before attempting to read it (for example,
|
|
for all of the channel map data in the ID header).
|
|
Implementations would do well to validate the indices of the channel map, also,
|
|
to ensure they meet all of the restrictions outlined in
|
|
<xref target="channel_mapping"/>, in order to avoid attempting to read data
|
|
from channels that do not exist.
|
|
</t>
|
|
|
|
<t>
|
|
To avoid excessive resource usage, we advise implementations to be especially
|
|
wary of streams that might cause them to process far more data than was
|
|
actually transmitted.
|
|
For example, a relatively small comment header may contain values for the
|
|
string lengths or user comment list length that imply that it is many
|
|
gigabytes in size.
|
|
Even computing the size of the required buffer could overflow a 32-bit integer,
|
|
and actually attempting to allocate such a buffer before verifying it would be
|
|
a reasonable size is a bad idea.
|
|
After reading the user comment list length, implementations might wish to
|
|
verify that the header contains at least the minimum amount of data for that
|
|
many comments (4 additional octets per comment, to indicate each has a
|
|
length of zero) before proceeding any further, again taking care to avoid
|
|
overflow in these calculations.
|
|
If allocating an array of pointers to point at these strings, the size of the
|
|
pointers may be larger than 4 octets, potentially requiring a separate
|
|
overflow check.
|
|
</t>
|
|
|
|
<t>
|
|
Another bug in this class we have observed more than once involves the handling
|
|
of invalid data at the end of a stream.
|
|
Often, implementations will seek to the end of a stream to locate the last
|
|
timestamp in order to compute its total duration.
|
|
If they do not find a valid capture pattern and Ogg page from the desired
|
|
logical stream, they will back up and try again.
|
|
If care is not taken to avoid re-scanning data that was already scanned, this
|
|
search can quickly devolve into something with a complexity that is quadratic
|
|
in the amount of invalid data.
|
|
</t>
|
|
|
|
<t>
|
|
In general when seeking, implementations will wish to be cautious about the
|
|
effects of invalid granule position values, and ensure all algorithms will
|
|
continue to make progress and eventually terminate, even if these are missing
|
|
or out-of-order.
|
|
</t>
|
|
|
|
<t>
|
|
Like most other container formats, Ogg Opus streams SHOULD NOT be used with
|
|
insecure ciphers or cipher modes that are vulnerable to known-plaintext
|
|
attacks.
|
|
Elements such as the Ogg page capture pattern and the magic signatures in the
|
|
ID header and the comment header all have easily predictable values, in
|
|
addition to various elements of the codec data itself.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="content_type" title="Content Type">
|
|
<t>
|
|
An "Ogg Opus file" consists of one or more sequentially multiplexed segments,
|
|
each containing exactly one Ogg Opus stream.
|
|
The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg".
|
|
</t>
|
|
|
|
<t>
|
|
If more specificity is desired, one MAY indicate the presence of Opus streams
|
|
using the codecs parameter defined in <xref target="RFC6381"/> and
|
|
<xref target="RFC5334"/>, e.g.,
|
|
</t>
|
|
<figure>
|
|
<artwork align="center"><![CDATA[
|
|
audio/ogg; codecs=opus
|
|
]]></artwork>
|
|
</figure>
|
|
<t>
|
|
for an Ogg Opus file.
|
|
</t>
|
|
|
|
<t>
|
|
The RECOMMENDED filename extension for Ogg Opus files is '.opus'.
|
|
</t>
|
|
|
|
<t>
|
|
When Opus is concurrently multiplexed with other streams in an Ogg container,
|
|
one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg"
|
|
mime-types, as defined in <xref target="RFC5334"/>.
|
|
Such streams are not strictly "Ogg Opus files" as described above,
|
|
since they contain more than a single Opus stream per sequentially
|
|
multiplexed segment, e.g. video or multiple audio tracks.
|
|
In such cases the the '.opus' filename extension is NOT RECOMMENDED.
|
|
</t>
|
|
|
|
<t>
|
|
In either case, this document updates <xref target="RFC5334"/>
|
|
to add 'opus' as a codecs parameter value with char[8]: 'OpusHead'
|
|
as Codec Identifier.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="iana" title="IANA Considerations">
|
|
<t>
|
|
This document updates the IANA Media Types registry to add .opus
|
|
as a file extension for "audio/ogg", and to add itself as a reference
|
|
alongside <xref target="RFC5334"/> for "audio/ogg", "video/ogg", and
|
|
"application/ogg" Media Types.
|
|
</t>
|
|
<t>
|
|
This document defines a new registry "Opus Channel Mapping Families" to
|
|
indicate how the semantic meanings of the channels in a multi-channel Opus
|
|
stream are described.
|
|
IANA is requested to create a new name space of "Opus Channel Mapping
|
|
Families".
|
|
This will be a new registry on the IANA Matrix, and not a subregistry of an
|
|
existing registry.
|
|
Modifications to this registry follow the "Specification Required" registration
|
|
policy as defined in <xref target="RFC5226"/>.
|
|
Each registry entry consists of a Channel Mapping Family Number, which is
|
|
specified in decimal in the range 0 to 255, inclusive, and a Reference (or
|
|
list of references)
|
|
Each Reference must point to sufficient documentation to describe what
|
|
information is coded in the Opus identification header for this channel
|
|
mapping family, how a demuxer determines the Stream Count ('N') and Coupled
|
|
Stream Count ('M') from this information, and how it determines the proper
|
|
interpretation of each of the decoded channels.
|
|
</t>
|
|
<t>
|
|
This document defines three initial assignments for this registry.
|
|
</t>
|
|
<texttable>
|
|
<ttcol>Value</ttcol><ttcol>Reference</ttcol>
|
|
<c>0</c><c>[RFCXXXX] <xref target="channel_mapping_0"/></c>
|
|
<c>1</c><c>[RFCXXXX] <xref target="channel_mapping_1"/></c>
|
|
<c>255</c><c>[RFCXXXX] <xref target="channel_mapping_255"/></c>
|
|
</texttable>
|
|
<t>
|
|
The designated expert will determine if the Reference points to a specification
|
|
that meets the requirements for permanence and ready availability laid out
|
|
in <xref target="RFC5226"/> and that it specifies the information
|
|
described above with sufficient clarity to allow interoperable
|
|
implementations.
|
|
</t>
|
|
</section>
|
|
|
|
<section anchor="Acknowledgments" title="Acknowledgments">
|
|
<t>
|
|
Thanks to Ben Campbell, Joel M. Halpern, Mark Harris, Greg Maxwell,
|
|
Christopher "Monty" Montgomery, Jean-Marc Valin, Stephan Wenger, and Mo Zanaty
|
|
for their valuable contributions to this document.
|
|
Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penquerc'h for
|
|
their feedback based on early implementations.
|
|
</t>
|
|
</section>
|
|
|
|
<section title="RFC Editor Notes">
|
|
<t>
|
|
In <xref target="iana"/>, "RFCXXXX" is to be replaced with the RFC number
|
|
assigned to this draft.
|
|
</t>
|
|
</section>
|
|
|
|
</middle>
|
|
<back>
|
|
<references title="Normative References">
|
|
&rfc2119;
|
|
&rfc3533;
|
|
&rfc3629;
|
|
&rfc5226;
|
|
&rfc5334;
|
|
&rfc6381;
|
|
&rfc6716;
|
|
|
|
<reference anchor="EBU-R128" target="https://tech.ebu.ch/loudness">
|
|
<front>
|
|
<title>Loudness Recommendation EBU R128</title>
|
|
<author>
|
|
<organization>EBU Technical Committee</organization>
|
|
</author>
|
|
<date month="August" year="2011"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="vorbis-comment"
|
|
target="https://www.xiph.org/vorbis/doc/v-comment.html">
|
|
<front>
|
|
<title>Ogg Vorbis I Format Specification: Comment Field and Header
|
|
Specification</title>
|
|
<author initials="C." surname="Montgomery"
|
|
fullname="Christopher "Monty" Montgomery"/>
|
|
<date month="July" year="2002"/>
|
|
</front>
|
|
</reference>
|
|
|
|
</references>
|
|
|
|
<references title="Informative References">
|
|
|
|
<!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml"?-->
|
|
&rfc4732;
|
|
&rfc6982;
|
|
&rfc7587;
|
|
|
|
<reference anchor="flac"
|
|
target="https://xiph.org/flac/format.html">
|
|
<front>
|
|
<title>FLAC - Free Lossless Audio Codec Format Description</title>
|
|
<author initials="J." surname="Coalson" fullname="Josh Coalson"/>
|
|
<date month="January" year="2008"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="hanning"
|
|
target="https://en.wikipedia.org/w/index.php?title=Window_function&oldid=703074467#Hann_.28Hanning.29_window">
|
|
<front>
|
|
<title>Hann window</title>
|
|
<author>
|
|
<organization>Wikipedia</organization>
|
|
</author>
|
|
<date month="February" year="2016"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="linear-prediction"
|
|
target="https://en.wikipedia.org/w/index.php?title=Linear_predictive_coding&oldid=687498962">
|
|
<front>
|
|
<title>Linear Predictive Coding</title>
|
|
<author>
|
|
<organization>Wikipedia</organization>
|
|
</author>
|
|
<date month="October" year="2015"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="lpc-sample"
|
|
target="https://svn.xiph.org/trunk/vorbis/lib/lpc.c">
|
|
<front>
|
|
<title>Autocorrelation LPC coeff generation algorithm
|
|
(Vorbis source code)</title>
|
|
<author initials="J." surname="Degener" fullname="Jutta Degener"/>
|
|
<author initials="C." surname="Bormann" fullname="Carsten Bormann"/>
|
|
<date month="November" year="1994"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="q-notation"
|
|
target="https://en.wikipedia.org/w/index.php?title=Q_%28number_format%29&oldid=697252615">
|
|
<front>
|
|
<title>Q (number format)</title>
|
|
<author><organization>Wikipedia</organization></author>
|
|
<date month="December" year="2015"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="replay-gain"
|
|
target="https://wiki.xiph.org/VorbisComment#Replay_Gain">
|
|
<front>
|
|
<title>VorbisComment: Replay Gain</title>
|
|
<author initials="C." surname="Parker" fullname="Conrad Parker"/>
|
|
<author initials="M." surname="Leese" fullname="Martin Leese"/>
|
|
<date month="June" year="2009"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="seeking"
|
|
target="https://wiki.xiph.org/Seeking">
|
|
<front>
|
|
<title>Granulepos Encoding and How Seeking Really Works</title>
|
|
<author initials="S." surname="Pfeiffer" fullname="Silvia Pfeiffer"/>
|
|
<author initials="C." surname="Parker" fullname="Conrad Parker"/>
|
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<author initials="G." surname="Maxwell" fullname="Greg Maxwell"/>
|
|
<date month="May" year="2012"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="vorbis-mapping"
|
|
target="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">
|
|
<front>
|
|
<title>The Vorbis I Specification, Section 4.3.9 Output Channel Order</title>
|
|
<author initials="C." surname="Montgomery"
|
|
fullname="Christopher "Monty" Montgomery"/>
|
|
<date month="January" year="2010"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="vorbis-trim"
|
|
target="https://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-132000A.2">
|
|
<front>
|
|
<title>The Vorbis I Specification, Appendix A: Embedding Vorbis
|
|
into an Ogg stream</title>
|
|
<author initials="C." surname="Montgomery"
|
|
fullname="Christopher "Monty" Montgomery"/>
|
|
<date month="November" year="2008"/>
|
|
</front>
|
|
</reference>
|
|
|
|
<reference anchor="wave-multichannel"
|
|
target="http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx">
|
|
<front>
|
|
<title>Multiple Channel Audio Data and WAVE Files</title>
|
|
<author>
|
|
<organization>Microsoft Corporation</organization>
|
|
</author>
|
|
<date month="March" year="2007"/>
|
|
</front>
|
|
</reference>
|
|
|
|
</references>
|
|
|
|
</back>
|
|
</rfc>
|