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RFC 7845

Proposed STD
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Ogg Encapsulation for the Opus Audio Codec

Part 1 of 2, p. 1 to 12
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Updates:    5334


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Internet Engineering Task Force (IETF)                     T. Terriberry
Request for Comments: 7845                           Mozilla Corporation
Updates: 5334                                                     R. Lee
Category: Standards Track                                    Voicetronix
ISSN: 2070-1721                                                 R. Giles
                                                     Mozilla Corporation
                                                              April 2016


               Ogg Encapsulation for the Opus Audio Codec

Abstract

   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.

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7845.

Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Packet Organization . . . . . . . . . . . . . . . . . . . . .   4
   4.  Granule Position  . . . . . . . . . . . . . . . . . . . . . .   6
     4.1.  Repairing Gaps in Real-Time Streams . . . . . . . . . . .   7
     4.2.  Pre-skip  . . . . . . . . . . . . . . . . . . . . . . . .   9
     4.3.  PCM Sample Position . . . . . . . . . . . . . . . . . . .   9
     4.4.  End Trimming  . . . . . . . . . . . . . . . . . . . . . .  10
     4.5.  Restrictions on the Initial Granule Position  . . . . . .  10
     4.6.  Seeking and Pre-roll  . . . . . . . . . . . . . . . . . .  11
   5.  Header Packets  . . . . . . . . . . . . . . . . . . . . . . .  12
     5.1.  Identification Header . . . . . . . . . . . . . . . . . .  12
       5.1.1.  Channel Mapping . . . . . . . . . . . . . . . . . . .  16
     5.2.  Comment Header  . . . . . . . . . . . . . . . . . . . . .  22
       5.2.1.  Tag Definitions . . . . . . . . . . . . . . . . . . .  25
   6.  Packet Size Limits  . . . . . . . . . . . . . . . . . . . . .  26
   7.  Encoder Guidelines  . . . . . . . . . . . . . . . . . . . . .  27
     7.1.  LPC Extrapolation . . . . . . . . . . . . . . . . . . . .  28
     7.2.  Continuous Chaining . . . . . . . . . . . . . . . . . . .  28
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  29
   9.  Content Type  . . . . . . . . . . . . . . . . . . . . . . . .  30
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  31
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  32
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  32
     11.2.  Informative References . . . . . . . . . . . . . . . . .  33
   Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . .  34
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  35

1.  Introduction

   The IETF Opus codec is a low-latency audio codec optimized for both
   voice and general-purpose audio.  See [RFC6716] for technical
   details.  This document defines the encapsulation of Opus in a
   continuous, logical Ogg bitstream [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.

   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

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   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.

   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
   [RFC3533].

2.  Terminology

   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
   [RFC2119].

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3.  Packet Organization

   An Ogg Opus stream is organized as follows (see Figure 1 for an
   example).

        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'

    Figure 1: Example Packet Organization for a Logical Ogg Opus Stream

   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 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.

   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 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.

   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
   Section 5.1.1), and is fixed over the entire length of the logical
   Ogg bitstream.

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   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 [RFC6716].  The remaining Opus packet is packed at the
   end of the Ogg packet using the regular, undelimited framing from
   Section 3 of [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.

   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 [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.

   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:

   o  the previous page with packet data does not end in a continued
      packet (does not end with a lacing value of 255) OR

   o  the page sequence numbers are not consecutive,

   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 [RFC6716].

   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

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   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:

   o  the next page with packet data does not have the 'continued
      packet' flag set, OR

   o  there is no next page with packet data, OR

   o  the page sequence numbers are not consecutive,

   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'.

4.  Granule Position

   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.

   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 Section 4.5.

   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.

   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.

   The duration of an Opus packet as defined in [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

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   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.

   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.

4.1.  Repairing Gaps in Real-Time Streams

   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.

   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 [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 [RFC6716].

   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).

   [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-

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   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 [RFC6716].

   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.

   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.

   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.

   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.

   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.

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4.2.  Pre-skip

   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.

   A 'pre-skip' field in the ID header (see 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.

   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.

   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.

4.3.  PCM Sample Position

   The PCM sample position is determined from the granule position using
   the following formula:

          'PCM sample position' = 'granule position' - 'pre-skip'

   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.

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   This can be converted into a playback time using the following
   formula:

                                    'PCM sample position'
                  'playback time' = ---------------------
                                           48000.0

   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.

   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
   [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.

4.4.  End Trimming

   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.

4.5.  Restrictions on the Initial Granule Position

   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

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   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.

   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.

   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.

4.6.  Seeking and Pre-roll

   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 [SEEKING] for an example
   of general implementation guidance.

   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

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   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).



(page 12 continued on part 2)

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