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RFC 7845


Ogg Encapsulation for the Opus Audio Codec

Part 2 of 2, p. 12 to 35
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5.  Header Packets

   An Ogg Opus logical stream contains exactly two mandatory header
   packets: an identification header and a comment header.

5.1.  Identification Header

      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...               :
     |                                                               |

                        Figure 2: ID Header Packet

   The fields in the identification (ID) header have the following

   1.  Magic Signature:

       This is an 8-octet (64-bit) field that allows codec
       identification and is human readable.  It contains, in order, the
       magic numbers:

          0x4F 'O'

          0x70 'p'

          0x75 'u'

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          0x73 's'

          0x48 'H'

          0x65 'e'

          0x61 'a'

          0x64 'd'

       Starting with "Op" helps distinguish it from audio data packets,
       as this is an invalid TOC sequence.

   2.  Version (8 bits, unsigned):

       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.

   3.  Output Channel Count 'C' (8 bits, unsigned):

       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 Section 5.1.1 for details.

   4.  Pre-skip (16 bits, unsigned, little endian):

       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.

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   5.  Input Sample Rate (32 bits, unsigned, little endian):

       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.

       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.

       An Ogg Opus player SHOULD select the playback sample rate
       according to the following procedure:

       1.  If the hardware supports 48 kHz playback, decode at 48 kHz.

       2.  Otherwise, if the hardware's highest available sample rate is
           a supported rate, decode at this sample rate.

       3.  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.

       4.  Otherwise, decode at 48 kHz and resample.

       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.

       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.

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   6.  Output Gain (16 bits, signed, little endian):

       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 [Q-NOTATION]).

       To apply the gain, an implementation could use the following:

                 sample *= pow(10, output_gain/(20.0*256))

       where 'output_gain' is the raw 16-bit value from the header.

       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 Section 5.2), the
       adjustment MUST be applied in addition to this output gain in
       order to achieve playback at the normalized volume.

       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.

       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.

   7.  Channel Mapping Family (8 bits, unsigned):

       This octet indicates the order and semantic meaning of the output

       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 Section 5.1.1.

   8.  Channel Mapping Table:

       This table defines the mapping from encoded streams to output
       channels.  Its contents are specified in Section 5.1.1.

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   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.

5.1.1.  Channel Mapping

   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.

      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...               :

                      Figure 3: Channel Mapping Table

   The fields in the channel mapping table have the following meaning:

   1.  Stream Count 'N' (8 bits, unsigned):

       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 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.

       For channel mapping family 0, this value defaults to 1, and is
       not coded.

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   2.  Coupled Stream Count 'M' (8 bits, unsigned):

       This is the number of streams whose decoders are to be configured
       to produce two channels (stereo).  This MUST be no larger than
       the total number of streams, N.

       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.

       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.

       For channel mapping family 0, this value defaults to (C - 1)
       (i.e., 0 for mono and 1 for stereo), and is not coded.

   3.  Channel Mapping (8*C bits):

       This contains one octet per output channel, indicating which
       decoded channel is to be used for each one.  Let 'index' be the
       value of this octet for a particular output channel.  This value
       MUST either be smaller than (M + N) or be the special value 255.
       If 'index' is less than 2*M, the output MUST be taken from
       decoding stream ('index'/2) as stereo and selecting the left
       channel if 'index' is even, and the right channel if 'index' is
       odd.  If 'index' is 2*M or larger, but less than 255, the output
       MUST be taken from decoding stream ('index' - M) as mono.  If
       'index' is 255, the corresponding output channel MUST contain
       pure silence.

       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.

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       For channel mapping family 0, the first index defaults to 0, and
       if C == 2, the second index defaults to 1.  Neither index is

   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.  Channel Mapping Family 0

   Allowed numbers of channels: 1 or 2.  RTP mapping.  This is the same
   channel interpretation as [RFC7587].

   o  1 channel: monophonic (mono).

   o  2 channels: stereo (left, right).

   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.  Channel Mapping Family 1

   Allowed numbers of channels: 1...8.  Vorbis channel order (see

   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

   o  1 channel: monophonic (mono).

   o  2 channels: stereo (left, right).

   o  3 channels: linear surround (left, center, right).

   o  4 channels: quadraphonic (front left, front right, rear left,
      rear right).

   o  5 channels: 5.0 surround (front left, front center, front right,
      rear left, rear right).

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   o  6 channels: 5.1 surround (front left, front center, front right,
      rear left, rear right, LFE).

   o  7 channels: 6.1 surround (front left, front center, front right,
      side left, side right, rear center, LFE).

   o  8 channels: 7.1 surround (front left, front center, front right,
      side left, side right, rear left, rear right, LFE).

   This set of surround options and speaker location orderings is the
   same as those used by the Vorbis codec [VORBIS-MAPPING].  The
   ordering is different from the one used by the WAVE
   [WAVE-MULTICHANNEL] and Free Lossless Audio Codec (FLAC) [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.  Channel Mapping Family 255

   Allowed numbers of channels: 1...255.  No defined channel meaning.

   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.  Undefined Channel Mappings

   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.  Downmixing

   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.

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   Implementations MAY use the matrices in Figures 4 through 9 to
   implement downmixing from multichannel files using channel mapping
   family 1 (Section, 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.

   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.

   L output = ( 0.585786 * left + 0.414214 * center                    )
   R output = (                   0.414214 * center + 0.585786 * right )

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

                  Figure 4: Stereo Downmix Matrix for the
                      Linear Surround Channel Mapping

       /          \   /                                     \ / FL \
       | L output |   | 0.422650 0.000000 0.366025 0.211325 | | FR |
       | R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL |
       \          /   \                                     / \ RR /

   Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
   1/(1 + sqrt(3)/2 + 1/2) for normalization.

   Figure 5: Stereo Downmix Matrix for the Quadraphonic Channel Mapping

                                                               / 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 |
                                                               \    /

   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.

       Figure 6: Stereo Downmix Matrix for the 5.0 Surround Mapping

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                                                                   /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 |
   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

       Figure 7: Stereo Downmix Matrix for the 5.1 Surround Mapping

     /                                                                \
     | 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 |
     \                                                                /

   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 Section and the matrices above.

       Figure 8: Stereo Downmix Matrix for the 6.1 Surround Mapping

    /                                                                 \
    | .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 |
    | .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 |
    \                                                                 /

   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 Section and the
   matrices above.

       Figure 9: Stereo Downmix Matrix for the 7.1 Surround Mapping

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5.2.  Comment Header

      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                 |
     :                                                               :

                     Figure 10: Comment Header Packet

   The comment header consists of a 64-bit 'magic signature' field,
   followed by data in the same format as the [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.

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   1.  Magic Signature:

       This is an 8-octet (64-bit) field that allows codec
       identification and is human readable.  It contains, in order, the
       magic numbers:

          0x4F 'O'

          0x70 'p'

          0x75 'u'

          0x73 's'

          0x54 'T'

          0x61 'a'

          0x67 'g'

          0x73 's'

       Starting with "Op" helps distinguish it from audio data packets,
       as this is an invalid TOC sequence.

   2.  Vendor String Length (32 bits, unsigned, little endian):

       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.

   3.  Vendor String (variable length, UTF-8 vector):

       This is a simple human-readable tag for vendor information,
       encoded as a UTF-8 string [RFC3629].  No terminating null octet
       is necessary.

       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.

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   4.  User Comment List Length (32 bits, unsigned, little endian):

       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.

   5.  User Comment #i String Length (32 bits, unsigned, little endian):

       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.

   6.  User Comment #i String (variable length, UTF-8 vector):

       This field contains a single user comment encoded as a UTF-8
       string [RFC3629].  There is one for each user comment indicated
       by the 'user comment list length' field.

   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.

   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.

   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

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   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

5.2.1.  Tag Definitions

   The user comment strings follow the NAME=value format described by
   [VORBIS-COMMENT] with the same recommended tag names: ARTIST, TITLE,
   DATE, ALBUM, and so on.

   Two new comment tags are introduced here:

   First, an optional gain for track normalization:


   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 [REPLAY-GAIN], except that the normal volume reference is the
   [EBU-R128] standard.

   Second, an optional gain for album normalization:


   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

   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.

   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

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   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.

   To avoid confusion with multiple normalization schemes, an Opus
   comment header SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN,
   REPLAYGAIN_ALBUM_PEAK tags, unless they are only to be used in some
   context where there is guaranteed to be no such confusion.
   [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.

6.  Packet Size Limits

   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.

   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

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   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.

   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.

   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.

7.  Encoder Guidelines

   When encoding Opus streams, Ogg muxers SHOULD take into account the
   algorithmic delay of the Opus encoder.

   In encoders derived from the reference implementation [RFC6716], the
   number of samples can be queried with

    opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples));

   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 [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.

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7.1.  LPC Extrapolation

   The first step in LPC extrapolation is to compute linear prediction
   coefficients [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.

   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

                         x(n) = \   a(k)*x(n - k)
                               k = 1

   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 [HANNING].

7.2.  Continuous Chaining

   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:

   1.  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.

   2.  Set the granule position of the last page to a point near the end
       of the last frame.

   3.  Begin the second segment with a copy of the last frame of the
       first segment.

   4.  Set the 'pre-skip' value of the second stream in such a way as to
       properly join the two streams.

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   5.  Continue the encoding process normally from there, without any
       reset to the encoder.

   In encoders derived from the reference implementation, inter-frame
   prediction can be turned off by calling

     opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));

   For best results, this implementation requires that prediction be
   explicitly enabled again before resuming normal encoding, even after
   a reset.

8.  Security Considerations

   Implementations of the Opus codec need to take appropriate security
   considerations into account, as outlined in [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

   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 Section 5.1.1, in order to avoid attempting
   to read data from channels that do not exist.

   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

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   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.

   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.

   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.

   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.

9.  Content Type

   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".

   If more specificity is desired, one MAY indicate the presence of Opus
   streams using the codecs parameter defined in [RFC6381] and
   [RFC5334], e.g.,

                            audio/ogg; codecs=opus

   for an Ogg Opus file.

   The RECOMMENDED filename extension for Ogg Opus files is '.opus'.

   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 [RFC5334].  Such streams
   are not strictly "Ogg Opus files" as described above, since they

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   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.

   In either case, this document updates [RFC5334] to add "opus" as a
   codecs parameter value with char[8]: 'OpusHead' as Codec Identifier.

10.  IANA Considerations

   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 [RFC5334] for "audio/ogg", "video/ogg", and
   "application/ogg" Media Types.

   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 [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.

   This document defines three initial assignments for this registry.

                   | Value | Reference                 |
                   | 0     | RFC 7845, Section |
                   |       |                           |
                   | 1     | RFC 7845, Section |
                   |       |                           |
                   | 255   | RFC 7845, Section |

   The designated expert will determine if the Reference points to a
   specification that meets the requirements for permanence and ready
   availability laid out in [RFC5226] and whether it specifies the
   information described above with sufficient clarity to allow
   interoperable implementations.

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11.  References

11.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3533]  Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
              RFC 3533, DOI 10.17487/RFC3533, May 2003,

   [RFC3629]  Yergeau, F., "UTF-8, a transformation format of ISO
              10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
              2003, <>.

   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              DOI 10.17487/RFC5226, May 2008,

   [RFC5334]  Goncalves, I., Pfeiffer, S., and C. Montgomery, "Ogg Media
              Types", RFC 5334, DOI 10.17487/RFC5334, September 2008,

   [RFC6381]  Gellens, R., Singer, D., and P. Frojdh, "The 'Codecs' and
              'Profiles' Parameters for "Bucket" Media Types", RFC 6381,
              DOI 10.17487/RFC6381, August 2011,

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
              September 2012, <>.

              EBU Technical Committee, "Loudness Recommendation EBU
              R128", August 2011, <>.

              Montgomery, C., "Ogg Vorbis I Format Specification:
              Comment Field and Header Specification", July 2002,

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11.2.  Informative References

   [RFC4732]  Handley, M., Ed., Rescorla, E., Ed., and IAB, "Internet
              Denial-of-Service Considerations", RFC 4732,
              DOI 10.17487/RFC4732, December 2006,

   [RFC7587]  Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
              for the Opus Speech and Audio Codec", RFC 7587,
              DOI 10.17487/RFC7587, June 2015,

   [FLAC]     Coalson, J., "FLAC - Free Lossless Audio Codec Format
              Description", January 2008,

   [HANNING]  Wikipedia, "Hann window", February 2016,

              Wikipedia, "Linear Predictive Coding", October 2015,

              Degener, J. and C. Bormann, "Autocorrelation LPC coeff
              generation algorithm (Vorbis source code)", November 1994,

              Wikipedia, "Q (number format)", December 2015,

              Parker, C. and M. Leese, "VorbisComment: Replay Gain",
              June 2009,

   [SEEKING]  Pfeiffer, S., Parker, C., and G. Maxwell, "Granulepos
              Encoding and How Seeking Really Works", May 2012,

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              Montgomery, C., "The Vorbis I Specification, Section 4.3.9
              Output Channel Order", January 2010,

              Montgomery, C., "The Vorbis I Specification, Appendix A:
              Embedding Vorbis into an Ogg stream", November 2008,

              Microsoft Corporation, "Multiple Channel Audio Data and
              WAVE Files", March 2007,


   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.

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Authors' Addresses

   Timothy B. Terriberry
   Mozilla Corporation
   331 E. Evelyn Ave.
   Mountain View, CA  94041
   United States

   Phone: +1 650 903-0800

   Ron Lee
   246 Pulteney Street, Level 1
   Adelaide, SA  5000

   Phone: +61 8 8232 9112

   Ralph Giles
   Mozilla Corporation
   163 West Hastings Street
   Vancouver, BC  V6B 1H5

   Phone: +1 778 785 1540