rfc7845.xml

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<!ENTITY RFC7587 SYSTEM "https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7587.xml">
]>
<rfc submissionType="IETF" number="7845" category="std" updates="5334" consensus="yes">
	<!-- Generated by id2xml 1.4.4 on 2019-02-23T10:34:17Z -->
	<?rfc compact="yes"?>
	<?rfc text-list-symbols="o*+-"?>
	<?rfc subcompact="no"?>
	<?rfc sortrefs="no"?>
	<?rfc symrefs="yes"?>
	<?rfc strict="yes"?>
	<?rfc toc="yes"?>
	<front>
	<title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title>
	<author fullname="Timothy B. Terriberry" initials="T." surname="Terriberry">
	<organization>Mozilla Corporation</organization>
	<address><postal><street>331 E. Evelyn Ave.</street>
	<street>Mountain View, CA  94041</street>
	<street>United States</street>
	</postal>
	<phone>+1 650 903-0800</phone>
	<email>tterribe@xiph.org</email>
	</address>
	</author>

	<author fullname="Ron Lee" initials="R." surname="Lee">
	<organization>Voicetronix</organization>
	<address><postal><street>246 Pulteney Street, Level 1</street>
	<street>Adelaide, SA  5000</street>
	<street>Australia</street>
	</postal>
	<phone>+61 8 8232 9112</phone>
	<email>ron@debian.org</email>
	</address>
	</author>

	<author fullname="Ralph Giles" initials="R." surname="Giles">
	<organization>Mozilla Corporation</organization>
	<address><postal><street>163 West Hastings Street</street>
	<street>Vancouver, BC  V6B 1H5</street>
	<street>Canada</street>
	</postal>
	<phone>+1 778 785 1540</phone>
	<email>giles@xiph.org</email>
	</address>
	</author>

	<date month="April" year="2016"/>
	<abstract><t>
   This document defines the Ogg encapsulation for the Opus interactive
   speech and audio codec.  This allows data encoded in the Opus format
   to be stored in an Ogg logical bitstream.</t>

	</abstract>
	</front>

	<middle>
	<section title="Introduction" anchor="section-1"><t>
   The IETF Opus codec is a low-latency audio codec optimized for both
   voice and general-purpose audio.  See <xref target="RFC6716"/> for technical
   details.  This document defines the encapsulation of Opus in a
   continuous, logical Ogg bitstream <xref target="RFC3533"/>.  Ogg encapsulation
   provides Opus with a long-term storage format supporting all of the
   essential features, including metadata, fast and accurate seeking,
   corruption detection, recapture after errors, low overhead, and the
   ability to multiplex Opus with other codecs (including video) with
   minimal buffering.  It also provides a live streamable format capable
   of delivery over a reliable stream-oriented transport, without
   requiring all the data (or even the total length of the data)
   up-front, in a form that is identical to the on-disk storage format.</t>

	<t>
   Ogg bitstreams are made up of a series of "pages", each of which
   contains data from one or more "packets".  Pages are the fundamental
   unit of multiplexing in an Ogg stream.  Each page is associated with
   a particular logical stream and contains a capture pattern and
   checksum, flags to mark the beginning and end of the logical stream,
   and a "granule position" that represents an absolute position in the
   stream, to aid seeking.  A single page can contain up to 65,025
   octets of packet data from up to 255 different packets.  Packets can
   be split arbitrarily across pages and continued from one page to the
   next (allowing packets much larger than would fit on a single page).
   Each page contains "lacing values" that indicate how the data is
   partitioned into packets, allowing a demultiplexer (demuxer) to
   recover the packet boundaries without examining the encoded data.  A
   packet is said to "complete" on a page when the page contains the
   final lacing value corresponding to that packet.</t>

	<t>
   This encapsulation defines the contents of the packet data, including
   the necessary headers, the organization of those packets into a
   logical stream, and the interpretation of the codec-specific granule
   position field.  It does not attempt to describe or specify the
   existing Ogg container format.  Readers unfamiliar with the basic
   concepts mentioned above are encouraged to review the details in
   <xref target="RFC3533"/>.</t>

	</section>

	<section title="Terminology" anchor="section-2"><t>
   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   <xref target="RFC2119"/>.</t>

	</section>

	<section title="Packet Organization" anchor="section-3"><t>
   An Ogg Opus stream is organized as follows (see Figure 1 for an
   example).</t>

	<figure title="Example Packet Organization for a Logical Ogg Opus Stream" anchor="ref-example-packet-organization-for-a-logical-ogg-opus-stream"><artwork><![CDATA[
     Page 0         Pages 1 ... n        Pages (n+1) ...
  +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
  |            | |   | |   |     |   | |           | |         | |
  |+----------+| |+-----------------+| |+-------------------+ +-----
  |||ID Header|| ||  Comment Header || ||Audio Data Packet 1| | ...
  |+----------+| |+-----------------+| |+-------------------+ +-----
  |            | |   | |   |     |   | |           | |         | |
  +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
  ^      ^                           ^
  |      |                           |
  |      |                           Mandatory Page Break
  |      |
  |      ID header is contained on a single page
  |
  'Beginning Of Stream'
]]></artwork>
	</figure>
	<t>
   There are two mandatory header packets.  The first packet in the
   logical Ogg bitstream MUST contain the identification (ID) header,
   which uniquely identifies a stream as Opus audio.  The format of this
   header is defined in <xref target="section-5.1"/>.  It is placed alone (without any
   other packet data) on the first page of the logical Ogg bitstream and
   completes on that page.  This page has its 'beginning of stream' flag
   set.</t>

	<t>
   The second packet in the logical Ogg bitstream MUST contain the
   comment header, which contains user-supplied metadata.  The format of
   this header is defined in <xref target="section-5.2"/>.  It MAY span multiple pages,
   beginning on the second page of the logical stream.  However many
   pages it spans, the comment header packet MUST finish the page on
   which it completes.</t>

	<t>
   All subsequent pages are audio data pages, and the Ogg packets they
   contain are audio data packets.  Each audio data packet contains one
   Opus packet for each of N different streams, where N is typically one
   for mono or stereo, but MAY be greater than one for multichannel
   audio.  The value N is specified in the ID header (see
   <xref target="section-5.1.1"/>), and is fixed over the entire length of the logical
   Ogg bitstream.</t>

	<t>
   The first (N - 1) Opus packets, if any, are packed one after another
   into the Ogg packet, using the self-delimiting framing from
   Appendix B of <xref target="RFC6716"/>.  The remaining Opus packet is packed at the
   end of the Ogg packet using the regular, undelimited framing from
   Section 3 of <xref target="RFC6716"/>.  All of the Opus packets in a single Ogg
   packet MUST be constrained to have the same duration.  An
   implementation of this specification SHOULD treat any Opus packet
   whose duration is different from that of the first Opus packet in an
   Ogg packet as if it were a malformed Opus packet with an invalid
   Table Of Contents (TOC) sequence.</t>

	<t>
   The TOC sequence at the beginning of each Opus packet indicates the
   coding mode, audio bandwidth, channel count, duration (frame size),
   and number of frames per packet, as described in Section 3.1
   of <xref target="RFC6716"/>.  The coding mode is one of SILK, Hybrid, or Constrained
   Energy Lapped Transform (CELT).  The combination of coding mode,
   audio bandwidth, and frame size is referred to as the configuration
   of an Opus packet.</t>

	<t>
   Packets are placed into Ogg pages in order until the end of stream.
   Audio data packets might span page boundaries.  The first audio data
   page could have the 'continued packet' flag set (indicating the first
   audio data packet is continued from a previous page) if, for example,
   it was a live stream joined mid-broadcast, with the headers pasted on
   the front.  If a page has the 'continued packet' flag set and one of
   the following conditions is also true:</t>

	<t><list style="symbols"><t>the previous page with packet data does not end in a continued
      packet (does not end with a lacing value of 255) OR</t>

	<t>the page sequence numbers are not consecutive,</t>

	</list>
	</t>

	<t>
   then a demuxer MUST NOT attempt to decode the data for the first
   packet on the page unless the demuxer has some special knowledge that
   would allow it to interpret this data despite the missing pieces.  An
   implementation MUST treat a zero-octet audio data packet as if it
   were a malformed Opus packet as described in Section 3.4
   of <xref target="RFC6716"/>.</t>

	<t>
   A logical stream ends with a page with the 'end of stream' flag set,
   but implementations need to be prepared to deal with truncated
   streams that do not have a page marked 'end of stream'.  There is no
   reason for the final packet on the last page to be a continued
   packet, i.e., for the final lacing value to be 255.  However,
   demuxers might encounter such streams, possibly as the result of a
   transfer that did not complete or of corruption.  If a packet</t>

	<t>
   continues onto a subsequent page (i.e., when the page ends with a
   lacing value of 255) and one of the following conditions is also
   true:</t>

	<t><list style="symbols"><t>the next page with packet data does not have the 'continued
      packet' flag set, OR</t>

	<t>there is no next page with packet data, OR</t>

	<t>the page sequence numbers are not consecutive,</t>

	</list>
	</t>

	<t>
   then a demuxer MUST NOT attempt to decode the data from that packet
   unless the demuxer has some special knowledge that would allow it to
   interpret this data despite the missing pieces.  There MUST NOT be
   any more pages in an Opus logical bitstream after a page marked 'end
   of stream'.</t>

	</section>

	<section title="Granule Position" anchor="section-4"><t>
   The granule position MUST be zero for the ID header page and the page
   where the comment header completes.  That is, the first page in the
   logical stream and the last header page before the first audio data
   page both have a granule position of zero.</t>

	<t>
   The granule position of an audio data page encodes the total number
   of PCM samples in the stream up to and including the last fully
   decodable sample from the last packet completed on that page.  The
   granule position of the first audio data page will usually be larger
   than zero, as described in <xref target="section-4.5"/>.</t>

	<t>
   A page that is entirely spanned by a single packet (that completes on
   a subsequent page) has no granule position, and the granule position
   field is set to the special value '-1' in two's complement.</t>

	<t>
   The granule position of an audio data page is in units of PCM audio
   samples at a fixed rate of 48 kHz (per channel; a stereo stream's
   granule position does not increment at twice the speed of a mono
   stream).  It is possible to run an Opus decoder at other sampling
   rates, but all Opus packets encode samples at a sampling rate that
   evenly divides 48 kHz.  Therefore, the value in the granule position
   field always counts samples assuming a 48 kHz decoding rate, and the
   rest of this specification makes the same assumption.</t>

	<t>
   The duration of an Opus packet as defined in <xref target="RFC6716"/> can be any
   multiple of 2.5 ms, up to a maximum of 120 ms.  This duration is
   encoded in the TOC sequence at the beginning of each packet.  The
   number of samples returned by a decoder corresponds to this duration
   exactly, even for the first few packets.  For example, a 20 ms packet
   fed to a decoder running at 48 kHz will always return 960 samples.  A
   demuxer can parse the TOC sequence at the beginning of each Ogg
   packet to work backwards or forwards from a packet with a known
   granule position (i.e., the last packet completed on some page) in
   order to assign granule positions to every packet, or even every
   individual sample.  The one exception is the last page in the stream,
   as described below.</t>

	<t>
   All other pages with completed packets after the first MUST have a
   granule position equal to the number of samples contained in packets
   that complete on that page plus the granule position of the most
   recent page with completed packets.  This guarantees that a demuxer
   can assign individual packets the same granule position when working
   forwards as when working backwards.  For this to work, there cannot
   be any gaps.</t>

	<section title="Repairing Gaps in Real-Time Streams" anchor="section-4.1"><t>
   In order to support capturing a real-time stream that has lost or not
   transmitted packets, a multiplexer (muxer) SHOULD emit packets that
   explicitly request the use of Packet Loss Concealment (PLC) in place
   of the missing packets.  Implementations that fail to do so still
   MUST NOT increment the granule position for a page by anything other
   than the number of samples contained in packets that actually
   complete on that page.</t>

	<t>
   Only gaps that are a multiple of 2.5 ms are repairable, as these are
   the only durations that can be created by packet loss or
   discontinuous transmission.  Muxers need not handle other gap sizes.
   Creating the necessary packets involves synthesizing a TOC byte
   (defined in Section 3.1 of <xref target="RFC6716"/>) -- and whatever additional
   internal framing is needed -- to indicate the packet duration for
   each stream.  The actual length of each missing Opus frame inside the
   packet is zero bytes, as defined in Section 3.2.1 of <xref target="RFC6716"/>.</t>

	<t>
   Zero-byte frames MAY be packed into packets using any of codes 0, 1,
   2, or 3.  When successive frames have the same configuration, the
   higher code packings reduce overhead.  Likewise, if the TOC
   configuration matches, the muxer MAY further combine the empty frames
   with previous or subsequent nonzero-length frames (using code 2 or
   variable bitrate (VBR) code 3).</t>

	<t>
   <xref target="RFC6716"/> does not impose any requirements on the PLC, but this
   section outlines choices that are expected to have a positive
   influence on most PLC implementations, including the reference
   implementation.  Synthesized TOC sequences SHOULD maintain the same
   mode, audio bandwidth, channel count, and frame size as the previous
   packet (if any).  This is the simplest and usually the most well-
   tested case for the PLC to handle and it covers all losses that do
   not include a configuration switch, as defined in Section 4.5
   of <xref target="RFC6716"/>.</t>

	<t>
   When a previous packet is available, keeping the audio bandwidth and
   channel count the same allows the PLC to provide maximum continuity
   in the concealment data it generates.  However, if the size of the
   gap is not a multiple of the most recent frame size, then the frame
   size will have to change for at least some frames.  Such changes
   SHOULD be delayed as long as possible to simplify things for PLC
   implementations.</t>

	<t>
   As an example, a 95 ms gap could be encoded as nineteen 5 ms frames
   in two bytes with a single constant bitrate (CBR) code 3 packet.  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.  The total bitrate of the
   latter approach, including Ogg overhead, is about 0.4 kbps, so the
   impact on file size is minimal.</t>

	<t>
   Changing modes is discouraged, since this causes some decoder
   implementations to reset their PLC state.  However, SILK and Hybrid
   mode frames cannot fill gaps that are not a multiple of 10 ms.  If
   switching to CELT mode is needed to match the gap size, a muxer
   SHOULD do so at the end of the gap to allow the PLC to function for
   as long as possible.</t>

	<t>
   In the example above, if the previous frame was a 20 ms SILK mode
   frame, the better solution is to synthesize a packet describing four
   20 ms SILK frames, followed by a packet with a single 10 ms SILK
   frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms
   gap.  This also requires four bytes to describe the synthesized
   packet data (two bytes for a CBR code 3 and one byte each for two
   code 0 packets) but three bytes of Ogg lacing overhead are needed to
   mark the packet boundaries.  At 0.6 kbps, this is still a minimal
   bitrate impact over a naive, low-quality solution.</t>

	<t>
   Since medium-band audio is an option only in the SILK mode, wideband
   frames SHOULD be generated if switching from that configuration to
   CELT mode, to ensure that any PLC implementation that does try to
   migrate state between the modes will be able to preserve all of the
   available audio bandwidth.</t>

	</section>

	<section title="Pre-skip" anchor="section-4.2"><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="section-5.1"/>) 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 title="PCM Sample Position" anchor="section-4.3"><t>
   The PCM sample position is determined from the granule position using
   the following formula:</t>

	<t><list hangIndent="7" style="hanging"><t>
          'PCM sample position' = 'granule position' - 'pre-skip'</t>

	</list>
	</t>

	<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 following
   formula:</t>

	<figure><artwork><![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 _after_ 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 title="End Trimming" anchor="section-4.4"><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 title="Restrictions on the Initial Granule Position" anchor="section-4.5"><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 title="Seeking and Pre-roll" anchor="section-4.6"><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 title="Header Packets" anchor="section-5"><t>
   An Ogg Opus logical stream contains exactly two mandatory header
   packets: an identification header and a comment header.</t>

	<section title="Identification Header" anchor="section-5.1"><figure title="ID Header Packet" anchor="ref-id-header-packet"><artwork><![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:</t>

	<t><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:
	<vspace blankLines="1"/>
	0x4F 'O'
	<vspace blankLines="1"/>
	0x70 'p'
	<vspace blankLines="1"/>
	0x75 'u'
	<vspace blankLines="1"/>
	0x73 's'
	<vspace blankLines="1"/>
	0x48 'H'
	<vspace blankLines="1"/>
	0x65 'e'
	<vspace blankLines="1"/>
	0x61 'a'
	<vspace blankLines="1"/>
	0x64 'd'
	<vspace blankLines="1"/>
	Starting with "Op" helps distinguish it from audio data packets,
       as this is an invalid TOC sequence.
	</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.
	</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="section-5.1.1"/> for details.
	</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.
	</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 _not_ 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.<vspace blankLines="1"/>
	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 that 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.
	</t>

	</list>
	</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"/>).
	<list style="hanging" hangIndent="10"><t hangText="To apply the gain, an implementation could use the following:">
	<vspace blankLines="1"/>
	sample *= pow(10, output_gain/(20.0*256))
	<vspace blankLines="1"/>
	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="section-5.2"/>), 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.
	</t>

	</list>
	</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="section-5.1.1"/>.
	</t>

	<t>Channel Mapping Table:<vspace blankLines="1"/>
	This table defines the mapping from encoded streams to output
       channels.  Its contents are specified in <xref target="section-5.1.1"/>.
	</t>

	</list>
	</t>

	<t>
   All fields in the ID headers are REQUIRED, except for '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 title="Channel Mapping" anchor="section-5.1.1"><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 Figure 3.</t>

	<figure title="Channel Mapping Table" anchor="ref-channel-mapping-table"><artwork><![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:</t>

	<t><list style="numbers"><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="section-3"/>.  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.
	</t>

	<t>Coupled Stream Count 'M' (8 bits, unsigned):<vspace blankLines="1"/>
	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.
	</t>

	<t>Channel Mapping (8*C bits):<vspace blankLines="1"/>
	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 title="Channel Mapping Family 0" anchor="section-5.1.1.1"><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>
	</t>

	<t>
   Special mapping: This channel mapping family also indicates that the
   content 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 title="Channel Mapping Family 1" anchor="section-5.1.1.2"><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.</t>

	<t><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 that prefer that terminology.</t>

	</section>

	<section title="Channel Mapping Family 255" anchor="section-5.1.1.3"><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 title="Undefined Channel Mappings" anchor="section-5.1.1.4"><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 title="Downmixing" anchor="section-5.1.1.5"><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 4 through 9 to
   implement downmixing from multichannel files using channel mapping
   family 1 (<xref target="section-5.1.1.2"/>), 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>

	<t><list style="hanging" hangIndent="68"><t hangText="L output = ( 0.585786 * left + 0.414214 * center">
	)
	<vspace blankLines="0"/>
	R output = (                   0.414214 * center + 0.585786 * right )
	</t>

	</list>
	</t>

	<t>
   Exact coefficient values are 1 and 1/sqrt(2), multiplied by
   1/(1 + 1/sqrt(2)) for normalization.</t>

	<t><list style="empty" hangIndent="15">
	<t><list style="hanging" hangIndent="4"><t hangText="Figure 4: Stereo Downmix Matrix for the">
	<vspace blankLines="0"/>
	Linear Surround Channel Mapping
	</t>

	</list>
	</t>

	</list>
	</t>

	<figure><artwork><![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>
	</figure>
	<t>
   Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
   1/(1 + sqrt(3)/2 + 1/2) for normalization.</t>

	<t><list style="hanging" hangIndent="-1"><t hangText="Figure 5: Stereo Downmix Matrix for the Quadraphonic Channel Mapping">
	<vspace blankLines="0"/>
	</t>

	</list>
	</t>

	<figure><artwork><![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>
	</figure>
	<t>
   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.</t>

	<t><list style="empty" hangIndent="4">
	<t><list style="hanging" hangIndent="-1"><t hangText="Figure 6: Stereo Downmix Matrix for the 5.0 Surround Mapping">
	<vspace blankLines="0"/>
	</t>

	</list>
	</t>

	</list>
	</t>

	<figure title="Stereo Downmix Matrix for the 5.1 Surround Mapping" anchor="ref-stereo-downmix-matrix-for-the-5.1-surround-mapping"><artwork><![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/
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.
]]></artwork>
	</figure>
	<t><list style="empty" hangIndent="2">
	<t><list style="empty"><t>     /                                                                \
     | 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 |
     \                                                                /</t>

	</list>
	</t>

	</list>
	</t>

	<t>
   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="section-5.1.1.2"/> and the matrices above.</t>

	<t><list hangIndent="4" style="hanging"><t>
       Figure 8: Stereo Downmix Matrix for the 6.1 Surround Mapping</t>

	</list>
	</t>

	<t><list style="empty" hangIndent="1">
	<t><list style="empty"><t>    /                                                                 \
    | .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 |
    | .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 |
    \                                                                 /</t>

	</list>
	</t>

	</list>
	</t>

	<t>
   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="section-5.1.1.2"/> and the
   matrices above.</t>

	<t><list hangIndent="4" style="hanging"><t>
       Figure 9: Stereo Downmix Matrix for the 7.1 Surround Mapping</t>

	</list>
	</t>

	</section>

	</section>

	</section>

	<section title="Comment Header" anchor="section-5.2"><figure title="Comment Header Packet" anchor="ref-comment-header-packet"><artwork><![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' field,
   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 specification is not present.</t>

	<t><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:
	<vspace blankLines="1"/>
	0x4F 'O'
	<vspace blankLines="1"/>
	0x70 'p'
	<vspace blankLines="1"/>
	0x75 'u'
	<vspace blankLines="1"/>
	0x73 's'
	<vspace blankLines="1"/>
	0x54 'T'
	<vspace blankLines="1"/>
	0x61 'a'
	<vspace blankLines="1"/>
	0x67 'g'
	<vspace blankLines="1"/>
	0x73 's'
	<vspace blankLines="1"/>
	Starting with "Op" helps distinguish it from audio data packets,
       as this is an invalid TOC sequence.
	</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.
	</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.
	</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.
	</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.
	</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' fields 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 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 that 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</t>

	<t>
   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 title="Tag Definitions" anchor="section-5.2.1"><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><artwork><![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><artwork><![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 _in addition_ to the 'output gain' value.  If a tool
   modifies the ID header's 'output gain' field, it MUST also update or</t>

	<t>
   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>

	</section>

	</section>

	<section title="Packet Size Limits" anchor="section-6"><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 VBR stream 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 title="Encoder Guidelines" anchor="section-7"><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>

	<t><list hangIndent="1" style="hanging"><t>
    opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&amp;delay_samples));</t>

	</list>
	</t>

	<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 title="LPC Extrapolation" anchor="section-7.1"><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><artwork><![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 title="Continuous Chaining" anchor="section-7.2"><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:</t>

	<t><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>

	<t><list hangIndent="2" style="hanging"><t>
     opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));</t>

	</list>
	</t>

	<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 title="Security Considerations" anchor="section-8"><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="section-5.1.1"/>, 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 signature' fields 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 title="Content Type" anchor="section-9"><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>

	<t><list hangIndent="25" style="hanging"><t>
                            audio/ogg; codecs=opus</t>

	</list>
	</t>

	<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</t>

	<t>
   contain more than a single Opus stream per sequentially multiplexed
   segment, e.g., video or multiple audio tracks.  In such cases, 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 title="IANA Considerations" anchor="section-10"><t>
   Per this document, IANA has updated the "Media Types" registry by
   adding .opus as a file extension for "audio/ogg" and adding 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 has created a new namespace
   of "Opus Channel Mapping Families".  This registry is listed on the
   IANA Matrix.  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><list style="hanging" hangIndent="-1"><t hangText="This document defines three initial assignments for this registry.">
	<vspace blankLines="0"/>
	</t>

	</list>
	</t>

	<texttable style="full"><ttcol> Value</ttcol>
	<ttcol> Reference</ttcol>
	<c>0</c>
	<c>RFC 7845, Section 5.1.1.1</c>
	<c>1</c>
	<c>RFC 7845, Section 5.1.1.2</c>
	<c>255</c>
	<c>RFC 7845, Section 5.1.1.3</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 whether it specifies the
   information described above with sufficient clarity to allow
   interoperable implementations.</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 fullname="C. Montgomery" initials="C." surname="Montgomery">
	</author>

	<date month="July" year="2002"/>
	</front>

	</reference>
	</references>
	<references title="Informative References">
	&RFC4732;
	&RFC7587;
	<reference anchor="FLAC" target="https://xiph.org/flac/format.html"><front>
	<title>FLAC - Free Lossless Audio Codec Format Description</title>
	<author fullname="J. Coalson" initials="J." surname="Coalson">
	</author>

	<date month="January" year="2008"/>
	</front>

	</reference>
	<reference anchor="HANNING" target="https://en.wikipedia.org/w/index"><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&amp;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 fullname="J. Degener" initials="J." surname="Degener">
	</author>

	<author fullname="C. Bormann" initials="C." surname="Bormann">
	</author>

	<date month="November" year="1994"/>
	</front>

	</reference>
	<reference anchor="Q-NOTATION" target="https://en.wikipedia.org/w/index.php?title=Q_%28number_format%29&amp;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 fullname="C. Parker" initials="C." surname="Parker">
	</author>

	<author fullname="M. Leese" initials="M." surname="Leese">
	</author>

	<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 fullname="S. Pfeiffer" initials="S." surname="Pfeiffer">
	</author>

	<author fullname="C. Parker" initials="C." surname="Parker">
	</author>

	<author fullname="G. Maxwell" initials="G." surname="Maxwell">
	</author>

	<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 fullname="C. Montgomery" initials="C." surname="Montgomery">
	</author>

	<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 fullname="C. Montgomery" initials="C." surname="Montgomery">
	</author>

	<date month="November" year="2008"/>
	</front>

	</reference>
	<reference anchor="WAVE-MULTICHANNEL" target="https://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>
	<section title="Acknowledgments" numbered="no" anchor="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>

	</back>

	</rfc>