ref: aad281878d650f680a895730eaa386df7cb3e8f0
dir: /doc/draft-ietf-codec-oggopus.xml/
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<rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-06">
<front>
<title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title>
<author initials="T.B." surname="Terriberry" fullname="Timothy B. Terriberry">
<organization>Mozilla Corporation</organization>
<address>
<postal>
<street>650 Castro Street</street>
<city>Mountain View</city>
<region>CA</region>
<code>94041</code>
<country>USA</country>
</postal>
<phone>+1 650 903-0800</phone>
<email>[email protected]</email>
</address>
</author>
<author initials="R." surname="Lee" fullname="Ron Lee">
<organization>Voicetronix</organization>
<address>
<postal>
<street>246 Pulteney Street, Level 1</street>
<city>Adelaide</city>
<region>SA</region>
<code>5000</code>
<country>Australia</country>
</postal>
<phone>+61 8 8232 9112</phone>
<email>[email protected]</email>
</address>
</author>
<author initials="R." surname="Giles" fullname="Ralph Giles">
<organization>Mozilla Corporation</organization>
<address>
<postal>
<street>163 West Hastings Street</street>
<city>Vancouver</city>
<region>BC</region>
<code>V6B 1H5</code>
<country>Canada</country>
</postal>
<phone>+1 778 785 1540</phone>
<email>[email protected]</email>
</address>
</author>
<date day="18" month="October" year="2014"/>
<area>RAI</area>
<workgroup>codec</workgroup>
<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.
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>
</abstract>
</front>
<middle>
<section anchor="intro" title="Introduction">
<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"/>.
</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 MAY 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 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 anchor="terminology" title="Terminology">
<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>
<t>
Implementations that fail to satisfy one or more "MUST" requirements are
considered non-compliant.
Implementations that satisfy all "MUST" requirements, but fail to satisfy one
or more "SHOULD" requirements are said to be "conditionally compliant".
All other implementations are "unconditionally compliant".
</t>
</section>
<section anchor="packet_organization" title="Packet Organization">
<t>
An Ogg Opus stream is organized as follows.
</t>
<t>
There are two mandatory header packets.
The granule position of the pages on which these packets complete MUST be zero.
</t>
<t>
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="id_header"/>.
It MUST be placed alone (without any other packet data) on the first page of
the logical Ogg bitstream, and MUST complete on that page.
This page MUST have 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="comment_header"/>.
It MAY span one or more 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="channel_mapping"/>), 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.
A decoder 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 TOC sequence.
</t>
<t>
The coding mode (SILK, Hybrid, or CELT), audio bandwidth, channel count,
duration (frame size), and number of frames per packet, are indicated in the
TOC (table of contents) sequence at the beginning of each Opus packet, as
described in Section 3.1 of <xref target="RFC6716"/>.
The combination of mode, audio bandwidth, and frame size is referred to as
the configuration of an Opus packet.
</t>
<t>
The first audio data page SHOULD NOT have the 'continued packet' flag set
(which would indicate the first audio data packet is continued from a previous
page).
Packets MUST be placed into Ogg pages in order until the end of stream.
Audio packets MAY span page boundaries.
A decoder 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>
The last page SHOULD have 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'.
The final packet on the last page SHOULD NOT be a continued packet, i.e., the
final lacing value SHOULD be less than 255.
There MUST NOT be any more pages in an Opus logical bitstream after a page
marked 'end of stream'.
</t>
</section>
<section anchor="granpos" title="Granule Position">
<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.
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 MUST
be 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 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 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 anchor="gap-repair" title="Repairing Gaps in Real-time Streams">
<t>
In order to support capturing a real-time stream that has lost or not
transmitted packets, a muxer SHOULD emit packets that explicitly request the
use of Packet Loss Concealment (PLC) in place of the missing packets.
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 non-zero-length frames (using
code 2 or 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 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 which does try to migrate state between
the modes will be able to preserve all of the available audio bandwidth.
</t>
</section>
<section anchor="preskip" title="Pre-skip">
<t>
There is some amount of latency introduced during the decoding process, to
allow for overlap in the CELT mode, stereo mixing in the SILK mode, and
resampling.
The encoder might have introduced additional latency through its own resampling
and analysis (though the exact amount is not specified).
Therefore, the first few samples produced by the decoder do not correspond to
real input audio, but are instead composed of padding inserted by the encoder
to compensate for this latency.
These samples need to be stored and decoded, as Opus is an asymptotically
convergent predictive codec, meaning the decoded contents of each frame depend
on the recent history of decoder inputs.
However, a decoder will want to skip these samples after decoding them.
</t>
<t>
A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals
the number of samples which SHOULD be skipped (decoded but discarded) at the
beginning of the stream.
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, and SHOULD NOT be played.
</t>
<t>
For example, if the first Opus frame uses the CELT mode, it will always
produce 120 samples of windowed overlap-add data.
However, the overlap data is initially all zeros (since there is no prior
frame), meaning this cannot, in general, accurately represent the original
audio.
The SILK mode requires additional delay to account for its analysis and
resampling latency.
The encoder delays the original audio to avoid this problem.
</t>
<t>
The pre-skip field MAY also be used to perform sample-accurate cropping of
already encoded streams.
In this case, a value of at least 3840 samples (80 ms) provides
sufficient history to the decoder that it will have converged
before the stream's output begins.
</t>
</section>
<section anchor="pcm_sample_position" title="PCM Sample Position">
<t>
<figure align="center">
<preamble>
The PCM sample position is determined from the granule position using the
formula
</preamble>
<artwork align="center"><![CDATA[
'PCM sample position' = 'granule position' - 'pre-skip' .
]]></artwork>
</figure>
</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.
<figure align="center">
<preamble>
This can be converted into a playback time using the formula
</preamble>
<artwork align="center"><![CDATA[
'PCM sample position'
'playback time' = --------------------- .
48000.0
]]></artwork>
</figure>
</t>
<t>
The initial PCM sample position before any samples are played is normally '0'.
In this case, the PCM sample position of the first audio sample to be played
starts at '1', because it marks the time on the clock
<spanx style="emph">after</spanx> that sample has been played, and a stream
that is exactly one second long has a final PCM sample position of '48000',
as in the example here.
</t>
<t>
Vorbis streams use a granule position smaller than the number of audio samples
contained in the first audio data page to indicate that some of those samples
are trimmed from the output (see <xref target="vorbis-trim"/>).
However, to do so, Vorbis requires that the first audio data page contains
exactly two packets, in order to allow the decoder to perform PCM position
adjustments before needing to return any PCM data.
Opus uses the pre-skip mechanism for this purpose instead, since the encoder
MAY introduce more than a single packet's worth of latency, and since very
large packets in streams with a very large number of channels might not fit
on a single page.
</t>
</section>
<section anchor="end_trimming" title="End Trimming">
<t>
The page with the 'end of stream' flag set MAY have a granule position that
indicates the page contains less audio data than would normally be returned by
decoding up through the final packet.
This is used to end the stream somewhere other than an even frame boundary.
The granule position of the most recent audio data page with completed packets
is used to make this determination, or '0' is used if there were no previous
audio data pages with a completed packet.
The difference between these granule positions indicates how many samples to
keep after decoding the packets that completed on the final page.
The remaining samples are discarded.
The number of discarded samples SHOULD be no larger than the number decoded
from the last packet.
</t>
</section>
<section anchor="start_granpos_restrictions"
title="Restrictions on the Initial Granule Position">
<t>
The granule position of the first audio data page with a completed packet MAY
be larger than the number of samples contained in packets that complete on
that page, however it MUST NOT be smaller, unless that page has the 'end of
stream' flag set.
Allowing a granule position larger than the number of samples allows the
beginning of a stream to be cropped or a live stream to be joined without
rewriting the granule position of all the remaining pages.
This means that the PCM sample position just before the first sample to be
played MAY be larger than '0'.
Synchronization when multiplexing with other logical streams still uses the PCM
sample position relative to '0' to compute sample times.
This does not affect the behavior of pre-skip: exactly 'pre-skip' samples
SHOULD be skipped from the beginning of the decoded output, even if the
initial PCM sample position is greater than zero.
</t>
<t>
On the other hand, a granule position that is smaller than the number of
decoded samples prevents a demuxer from working backwards to assign each
packet or each individual sample a valid granule position, since granule
positions are non-negative.
A decoder MUST reject as invalid any stream where 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 reject as invalid
any stream where its granule position is smaller than the 'pre-skip' amount.
This would indicate that there are more samples to be skipped from the initial
decoded output than exist in the stream.
If the granule position is smaller than the number of decoded samples produced
by the packets that complete on that page, then a demuxer MUST use an initial
granule position of '0', and can work forwards from '0' to timestamp
individual packets.
If the granule position is larger than the number of decoded samples available,
then the demuxer MUST still work backwards as described above, even if the
'end of stream' flag is set, to determine the initial granule position, and
thus the initial PCM sample position.
Both of these will be greater than '0' in this case.
</t>
</section>
<section anchor="seeking_and_preroll" title="Seeking and Pre-roll">
<t>
Seeking in Ogg files is best performed using a bisection search for a page
whose granule position corresponds to a PCM position at or before the seek
target.
With appropriately weighted bisection, accurate seeking can be performed with
just three or four bisections even in multi-gigabyte files.
See <xref target="seeking"/> for general implementation guidance.
</t>
<t>
When seeking within an Ogg Opus stream, the decoder 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, the decoder SHOULD start decoding from the beginning of the
stream, applying pre-skip as normal, regardless of whether the pre-skip is
larger or smaller than 80 ms, and then continue to discard samples
to reach the seek target (if any).
</t>
</section>
</section>
<section anchor="headers" title="Header Packets">
<t>
An Opus stream contains exactly two mandatory header packets:
an identification header and a comment header.
</t>
<section anchor="id_header" title="Identification Header">
<figure anchor="id_header_packet" title="ID Header Packet" align="center">
<artwork align="center"><![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'O' | 'p' | 'u' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'H' | 'e' | 'a' | 'd' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version = 1 | Channel Count | Pre-skip |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Input Sample Rate (Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Output Gain (Q7.8 in dB) | Mapping Family| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ :
| |
: Optional Channel Mapping Table... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]></artwork>
</figure>
<t>
The fields in the identification (ID) header have the following meaning:
<list style="numbers">
<t><spanx style="strong">Magic Signature</spanx>:
<vspace blankLines="1"/>
This is an 8-octet (64-bit) field that allows codec identification and is
human-readable.
It contains, in order, the magic numbers:
<list style="empty">
<t>0x4F 'O'</t>
<t>0x70 'p'</t>
<t>0x75 'u'</t>
<t>0x73 's'</t>
<t>0x48 'H'</t>
<t>0x65 'e'</t>
<t>0x61 'a'</t>
<t>0x64 'd'</t>
</list>
Starting with "Op" helps distinguish it from audio data packets, as this is an
invalid TOC sequence.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Version</spanx> (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, a decoder implementing this specification SHOULD accept any stream
with a version number of '15' or less, and SHOULD assume any stream with a
version number '16' or greater is incompatible.
The initial version '1' was chosen to keep implementations from relying on this
octet as a null terminator for the "OpusHead" string.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Output Channel Count</spanx> 'C' (8 bits, unsigned):
<vspace blankLines="1"/>
This is the number of output channels.
This might be different than the number of encoded channels, which can change
on a packet-by-packet basis.
This value MUST NOT be zero.
The maximum allowable value depends on the channel mapping family, and might be
as large as 255.
See <xref target="channel_mapping"/> for details.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Pre-skip</spanx> (16 bits, unsigned, little
endian):
<vspace blankLines="1"/>
This is the number of samples (at 48 kHz) to discard from the decoder
output when starting playback, and also the number to subtract from a page's
granule position to calculate its PCM sample position.
When cropping the beginning of existing Ogg Opus streams, a pre-skip of at
least 3,840 samples (80 ms) is RECOMMENDED to ensure complete
convergence in the decoder.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Input Sample Rate</spanx> (32 bits, unsigned, little
endian):
<vspace blankLines="1"/>
This field is <spanx style="emph">not</spanx> the sample rate to use for
playback of the encoded data.
<vspace blankLines="1"/>
Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and
20 kHz.
Each packet in the stream can have a different audio bandwidth.
Regardless of the audio bandwidth, the reference decoder supports decoding any
stream at a sample rate of 8, 12, 16, 24, or 48 kHz.
The original sample rate of the encoder input 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 highest supported rate above this and
resample.</t>
<t>Otherwise, decode at 48 kHz and resample.</t>
</list>
However, the 'Input Sample Rate' field allows the encoder 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, a non-player decoder writing PCM format samples to disk might
choose to resample the output 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 as the one they fed to the encoder.
<vspace blankLines="1"/>
A value of zero indicates 'unspecified'.
Encoders SHOULD write the actual input sample rate or zero, but decoder
implementations which do something with this field SHOULD take care to behave
sanely if given crazy values (e.g., do not actually upsample the output to
10 MHz if requested).
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Output Gain</spanx> (16 bits, signed, little
endian):
<vspace blankLines="1"/>
This is a gain to be applied by the decoder.
It is 20*log10 of the factor to scale the decoder output by 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).
<figure align="center">
<preamble>
To apply the gain, a decoder could use
</preamble>
<artwork align="center"><![CDATA[
sample *= pow(10, output_gain/(20.0*256)) ,
]]></artwork>
<postamble>
where output_gain is the raw 16-bit value from the header.
</postamble>
</figure>
<vspace blankLines="1"/>
Virtually all players and media frameworks SHOULD apply it by default.
If a player chooses to apply any volume adjustment or gain modification, such
as the R128_TRACK_GAIN (see <xref target="comment_header"/>), the adjustment
MUST be applied in addition to this output gain in order to achieve playback
at the normalized volume.
<vspace blankLines="1"/>
An encoder SHOULD set this field to zero, and instead apply any gain prior to
encoding, when this is possible and does not conflict with the user's wishes.
A nonzero output gain indicates the gain was adjusted after encoding, or that
a user wished to adjust the gain for playback while preserving the ability
to recover the original signal amplitude.
<vspace blankLines="1"/>
Although the output gain has enormous range (+/- 128 dB, enough to amplify
inaudible sounds to the threshold of physical pain), most applications can
only reasonably use a small portion of this range around zero.
The large range serves in part to ensure that gain can always be losslessly
transferred between OpusHead and R128 gain tags (see below) without
saturating.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Channel Mapping Family</spanx> (8 bits,
unsigned):
<vspace blankLines="1"/>
This octet indicates the order and semantic meaning of the output channels.
<vspace blankLines="1"/>
Each possible value of this octet indicates a mapping family, which defines a
set of allowed channel counts, and the ordered set of channel names for each
allowed channel count.
The details are described in <xref target="channel_mapping"/>.
</t>
<t><spanx style="strong">Channel Mapping Table</spanx>:
This table defines the mapping from encoded streams to output channels.
It is omitted when the channel mapping family is 0, but REQUIRED otherwise.
Its contents are specified in <xref target="channel_mapping"/>.
</t>
</list>
</t>
<t>
All fields in the ID headers are REQUIRED, except for the channel mapping
table, which is omitted when the channel mapping family is 0.
Implementations SHOULD reject ID headers which do not contain enough data for
these fields, even if they 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 reject it for containing additional data not
specified here.
However, implementations MAY reject streams in which the ID header does not
complete on the first page.
</t>
<section anchor="channel_mapping" title="Channel Mapping">
<t>
An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly
larger number of decoded channels (M+N) to yet another number of output
channels (C), which might be larger or smaller than the number of decoded
channels.
The order and meaning of these channels are defined by a channel mapping,
which consists of the 'channel mapping family' octet and, for channel mapping
families other than family 0, a channel mapping table, as illustrated in
<xref target="channel_mapping_table"/>.
</t>
<figure anchor="channel_mapping_table" title="Channel Mapping Table"
align="center">
<artwork align="center"><![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
| Stream Count |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Coupled Count | Channel Mapping... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
]]></artwork>
</figure>
<t>
The fields in the channel mapping table have the following meaning:
<list style="numbers" counter="8">
<t><spanx style="strong">Stream Count</spanx> 'N' (8 bits, unsigned):
<vspace blankLines="1"/>
This is the total number of streams encoded in each Ogg packet.
This value is necessary to correctly parse the packed Opus packets inside an
Ogg packet, as described in <xref target="packet_organization"/>.
This value MUST NOT be zero, as without at least one Opus packet with a valid
TOC sequence, a demuxer cannot recover the duration of an Ogg packet.
<vspace blankLines="1"/>
For channel mapping family 0, this value defaults to 1, and is not coded.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Coupled Stream Count</spanx> 'M' (8 bits, unsigned):
This is the number of streams whose decoders are to be configured to produce
two channels.
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 encoder input is not 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 first M Opus decoders are
to be initialized for stereo output, and the remaining N-M decoders are to be
initialized for mono only.
The total number of decoded channels, (M+N), MUST be no larger than 255, as
there is no way to index more channels than that in the channel mapping.
<vspace blankLines="1"/>
For channel mapping family 0, this value defaults to C-1 (i.e., 0 for mono
and 1 for stereo), and is not coded.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Channel Mapping</spanx> (8*C bits):
This contains one octet per output channel, indicating which decoded channel
is to be used for each one.
Let 'index' be the value of this octet for a particular output channel.
This value MUST either be smaller than (M+N), or be the special value 255.
If 'index' is less than 2*M, the output MUST be taken from decoding stream
('index'/2) as stereo and selecting the left channel if 'index' is even, and
the right channel if 'index' is odd.
If 'index' is 2*M or larger, but less than 255, the output MUST be taken from
decoding stream ('index'-M) as mono.
If 'index' is 255, the corresponding output channel MUST contain pure silence.
<vspace blankLines="1"/>
The number of output channels, C, is not constrained to match the number of
decoded channels (M+N).
A single index value MAY appear multiple times, i.e., the same decoded channel
might be mapped to multiple output channels.
Some decoded channels might not be assigned to any output channel, as well.
<vspace blankLines="1"/>
For channel mapping family 0, the first index defaults to 0, and if C==2,
the second index defaults to 1.
Neither index is coded.
</t>
</list>
</t>
<t>
After producing the output channels, the channel mapping family determines the
semantic meaning of each one.
There are three defined mapping families in this specification.
</t>
<section anchor="channel_mapping_0" title="Channel Mapping Family 0">
<t>
Allowed numbers of channels: 1 or 2.
RTP mapping.
</t>
<t>
<list style="symbols">
<t>1 channel: monophonic (mono).</t>
<t>2 channels: stereo (left, right).</t>
</list>
<spanx style="strong">Special mapping</spanx>: This channel mapping value also
indicates that the contents consists of a single Opus stream that is stereo if
and only if C==2, with stream index 0 mapped to output channel 0 (mono, or
left channel) and stream index 1 mapped to output channel 1 (right channel)
if stereo.
When the 'channel mapping family' octet has this value, the channel mapping
table MUST be omitted from the ID header packet.
</t>
</section>
<section anchor="channel_mapping_1" title="Channel Mapping Family 1">
<t>
Allowed numbers of channels: 1...8.
Vorbis channel order.
</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 indicies.
<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
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, often mapped to a subwoofer
with no particular spatial position.
Implementations SHOULD identify 'side' or 'rear' speaker locations with
'surround' and 'back' as appropriate when interfacing with audio formats
or systems which prefer that terminology.
</t>
</section>
<section anchor="channel_mapping_255"
title="Channel Mapping Family 255">
<t>
Allowed numbers of channels: 1...255.
No defined channel meaning.
</t>
<t>
Channels are unidentified.
General-purpose players SHOULD NOT attempt to play these streams, and offline
decoders MAY deinterleave the output into separate PCM files, one per channel.
Decoders SHOULD NOT produce output for channels mapped to stream index 255
(pure silence) unless they have no other way to indicate the index of
non-silent channels.
</t>
</section>
<section anchor="channel_mapping_undefined"
title="Undefined Channel Mappings">
<t>
The remaining channel mapping families (2...254) are reserved.
A decoder encountering a reserved channel mapping family value SHOULD act as
though the value is 255.
</t>
</section>
<section anchor="downmix" title="Downmixing">
<t>
An Ogg Opus player MUST play any Ogg Opus stream 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 following matricies to implement downmixing from
multichannel files using <xref target="channel_mapping_1">Channel Mapping
Family 1</xref>, which are known to give acceptable results for stereo.
Matricies for 3 and 4 channels are normalized so each coefficent 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 matricies the front left and front right channels are generally
passed through directly.
When a surround channel is split between both the left and right stereo
channels, coefficients are chosen so their squares sum to 1, which
helps preserve the perceived intensity.
Rear channels are mixed more diffusely or attenuated to maintain focus
on the front channels.
</t>
<figure anchor="downmix-matrix-3"
title="Stereo downmix matrix for the linear surround channel mapping"
align="center">
<artwork align="center"><![CDATA[
L output = ( 0.585786 * left + 0.414214 * center )
R output = ( 0.414214 * center + 0.585786 * right )
]]></artwork>
<postamble>
Exact coefficient values are 1 and 1/sqrt(2), multiplied by
1/(1 + 1/sqrt(2)) for normalization.
</postamble>
</figure>
<figure anchor="downmix-matrix-4"
title="Stereo downmix matrix for the quadraphonic channel mapping"
align="center">
<artwork align="center"><![CDATA[
/ \ / \ / FL \
| L output | | 0.422650 0.000000 0.366025 0.211325 | | FR |
| R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL |
\ / \ / \ RR /
]]></artwork>
<postamble>
Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
1/(1 + sqrt(3)/2 + 1/2) for normalization.
</postamble>
</figure>
<figure anchor="downmix-matrix-5"
title="Stereo downmix matrix for the 5.0 surround mapping"
align="center">
<artwork align="center"><![CDATA[
/ FL \
/ \ / \ | FC |
| L | | 0.650802 0.460186 0.000000 0.563611 0.325401 | | FR |
| R | = | 0.000000 0.460186 0.650802 0.325401 0.563611 | | RL |
\ / \ / | RR |
\ /
]]></artwork>
<postamble>
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2)
for normalization.
</postamble>
</figure>
<figure anchor="downmix-matrix-6"
title="Stereo downmix matrix for the 5.1 surround mapping"
align="center">
<artwork align="center"><![CDATA[
/FL \
/ \ / \ |FC |
|L| | 0.529067 0.374107 0.000000 0.458186 0.264534 0.374107 | |FR |
|R| = | 0.000000 0.374107 0.529067 0.264534 0.458186 0.374107 | |RL |
\ / \ / |RR |
\LFE/
]]></artwork>
<postamble>
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2))
for normalization.
</postamble>
</figure>
<figure anchor="downmix-matrix-7"
title="Stereo downmix matrix for the 6.1 surround mapping"
align="center">
<artwork align="center"><![CDATA[
/ \
| 0.455310 0.321953 0.000000 0.394310 0.227655 0.278819 0.321953 |
| 0.000000 0.321953 0.455310 0.227655 0.394310 0.278819 0.321953 |
\ /
]]></artwork>
<postamble>
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and
sqrt(3)/2/sqrt(2), multiplied by
2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 +
sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization.
The coeffients are in the same order as in <xref target="channel_mapping_1" />,
and the matricies above.
</postamble>
</figure>
<figure anchor="downmix-matrix-8"
title="Stereo downmix matrix for the 7.1 surround mapping"
align="center">
<artwork align="center"><![CDATA[
/ \
| .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 |
| .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 |
\ /
]]></artwork>
<postamble>
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
2/(2 + 2/sqrt(2) + sqrt(3)) for normalization.
The coeffients are in the same order as in <xref target="channel_mapping_1" />,
and the matricies above.
</postamble>
</figure>
</section>
</section> <!-- end channel_mapping_table -->
</section> <!-- end id_header -->
<section anchor="comment_header" title="Comment Header">
<figure anchor="comment_header_packet" title="Comment Header Packet"
align="center">
<artwork align="center"><![CDATA[
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'O' | 'p' | 'u' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'T' | 'a' | 'g' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
: Vendor String... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment List Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment #0 String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
: User Comment #0 String... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment #1 String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: :
]]></artwork>
</figure>
<t>
The comment header consists of a 64-bit magic signature, followed by data in
the same format as the <xref target="vorbis-comment"/> header used in Ogg
Vorbis, except (like Ogg Theora and Speex) the final "framing bit" specified
in the Vorbis spec is not present.
<list style="numbers">
<t><spanx style="strong">Magic Signature</spanx>:
<vspace blankLines="1"/>
This is an 8-octet (64-bit) field that allows codec identification and is
human-readable.
It contains, in order, the magic numbers:
<list style="empty">
<t>0x4F 'O'</t>
<t>0x70 'p'</t>
<t>0x75 'u'</t>
<t>0x73 's'</t>
<t>0x54 'T'</t>
<t>0x61 'a'</t>
<t>0x67 'g'</t>
<t>0x73 's'</t>
</list>
Starting with "Op" helps distinguish it from audio data packets, as this is an
invalid TOC sequence.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Vendor String Length</spanx> (32 bits, unsigned,
little endian):
<vspace blankLines="1"/>
This field gives the length of the following vendor string, in octets.
It MUST NOT indicate that the vendor string is longer than the rest of the
packet.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">Vendor String</spanx> (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 encoding applications can use the 'ENCODER' user comment tag
to identify themselves.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">User Comment List Length</spanx> (32 bits, unsigned,
little endian):
<vspace blankLines="1"/>
This field indicates the number of user-supplied comments.
It MAY indicate there are zero user-supplied comments, in which case there are
no additional fields in the packet.
It MUST NOT indicate that there are so many comments that the comment string
lengths would require more data than is available in the rest of the packet.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">User Comment #i String Length</spanx> (32 bits,
unsigned, little endian):
<vspace blankLines="1"/>
This field gives the length of the following user comment string, in octets.
There is one for each user comment indicated by the 'user comment list length'
field.
It MUST NOT indicate that the string is longer than the rest of the packet.
<vspace blankLines="1"/>
</t>
<t><spanx style="strong">User Comment #i String</spanx> (variable length, UTF-8
vector):
<vspace blankLines="1"/>
This field contains a single user comment string.
There is one for each user comment indicated by the 'user comment list length'
field.
</t>
</list>
</t>
<t>
The vendor string length and user comment list length are REQUIRED, and
implementations SHOULD reject comment headers that do not contain enough data
for these fields, or that do not contain enough data for the corresponding
vendor string or user comments they describe.
Making this check before allocating the associated memory to contain the data
helps prevent a possible Denial-of-Service (DoS) attack from small comment
headers that claim to contain strings longer than the entire packet or more
user comments than than could possibly fit in the packet.
</t>
<t>
Immediately following the user comment list, the comment header MAY
contain zero-padding or other binary data which is not specified here.
If the least-significant bit of the first byte of this data is 1, then editors
SHOULD preserve the contents of this data when updating the tags, but if this
bit is 0, all such data MAY be treated as padding, and truncated or discarded
as desired.
</t>
<section anchor="comment_format" title="Tag Definitions">
<t>
The user comment strings follow the NAME=value format described by
<xref target="vorbis-comment"/> with the same recommended tag names:
ARTIST, TITLE, DATE, ALBUM, and so on.
</t>
<t>
Two new comment tags are introduced here:
</t>
<figure align="center">
<preamble>An optional gain for track nomalization</preamble>
<artwork align="left"><![CDATA[
R128_TRACK_GAIN=-573
]]></artwork>
<postamble>
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.
</postamble>
</figure>
<t>
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>
<figure align="center">
<preamble>An optional gain for album nomalization</preamble>
<artwork align="left"><![CDATA[
R128_ALBUM_GAIN=111
]]></artwork>
<postamble>
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.
</postamble>
</figure>
<t>
An Ogg Opus stream MUST NOT have more than one of each tag, and if present
their values MUST be an integer from -32768 to 32767, inclusive,
represented in ASCII as a base 10 number with no whitespace.
A leading '+' or '-' character is valid.
Leading zeros are also permitted, but the value MUST be represented by
no more than 6 characters.
Other non-digit characters MUST NOT be present.
</t>
<t>
If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent
the R128 normalization gain relative to the 'output gain' field specified
in the ID header.
If a player chooses to make use of the R128_TRACK_GAIN tag or the
R128_ALBUM_GAIN tag, it MUST apply those gains
<spanx style="emph">in addition</spanx> to the 'output gain' value.
If a tool modifies the ID header's 'output gain' field, it MUST also update or
remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present.
An encoder SHOULD assume that by default tools will respect 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.
<xref target="EBU-R128"/> normalization is preferred to the earlier
REPLAYGAIN schemes because of its clear definition and adoption by industry.
Peak normalizations are difficult to calculate reliably for lossy codecs
because of variation in excursion heights due to decoder differences.
In the authors' investigations they were not applied consistently or broadly
enough to merit inclusion here.
</t>
</section> <!-- end comment_format -->
</section> <!-- end comment_header -->
</section> <!-- end headers -->
<section anchor="packet_size_limits" title="Packet Size Limits">
<t>
Technically, valid Opus packets can be arbitrarily large due to the padding
format, although the amount of non-padding data they can contain is bounded.
These packets might be spread over a similarly enormous number of Ogg pages.
Encoders SHOULD use no more padding than is necessary to make a variable
bitrate (VBR) stream constant bitrate (CBR).
Decoders SHOULD avoid attempting to allocate excessive amounts of memory when
presented with a very large packet.
The presence of an extremely large packet in the stream could indicate a
memory exhaustion attack or stream corruption.
Decoders SHOULD reject a packet that is too large to process, and display a
warning message.
</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, or about 60 kB per
Opus stream.
With 255 streams, this is 15,630,988 octets (14.9 MB) 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.
The largest packet consisting of entirely useful data is
(15,326*N - 2) octets, or about 15 kB per stream.
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, or about 7.5 kB
per stream.
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).
With N=8, the maximum number of channels currently defined by mapping
family 1, this gives a maximum packet size of 61,310 octets, or just
under 60 kB.
This is still quite conservative, as it assumes each output channel is taken
from one decoded channel of a stereo packet.
An implementation could reasonably choose any of these numbers for its internal
limits.
</t>
</section>
<section anchor="encoder" title="Encoder Guidelines">
<t>
When encoding Opus streams, Ogg muxers SHOULD take into account the
algorithmic delay of the Opus encoder.
</t>
<figure align="center">
<preamble>
In encoders derived from the reference implementation, the number of
samples can be queried with:
</preamble>
<artwork align="center"><![CDATA[
opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples));
]]></artwork>
</figure>
<t>
To achieve good quality in the very first samples of a stream, the Ogg encoder
MAY use linear predictive coding (LPC) extrapolation
<xref target="linear-prediction"/> to generate at least 120 extra samples at
the beginning to avoid the Opus encoder having to encode a discontinuous
signal.
For an input file containing 'length' samples, the Ogg encoder SHOULD set the
pre-skip header value to delay_samples+extra_samples, encode at least
length+delay_samples+extra_samples samples, and set the granulepos of the last
page to length+delay_samples+extra_samples.
This ensures that the encoded file has the same duration as the original, with
no time offset. The best way to pad the end of the stream is to also use LPC
extrapolation, but zero-padding is also acceptable.
</t>
<section anchor="lpc" title="LPC Extrapolation">
<t>
The first step in LPC extrapolation is to compute linear prediction
coefficients. <xref target="lpc-sample"/>
When extending the end of the signal, order-N (typically with N ranging from 8
to 40) LPC analysis is performed on a window near the end of the signal.
The last N samples are used as memory to an infinite impulse response (IIR)
filter.
</t>
<figure align="center">
<preamble>
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:
</preamble>
<artwork align="center"><![CDATA[
N
---
x(n) = \ a(k)*x(n-k)
/
---
k=1
]]></artwork>
</figure>
<t>
The process is repeated independently for each channel.
It is possible to extend the beginning of the signal by applying the same
process backward in time.
When extending the beginning of the signal, it is best to apply a "fade in" to
the extrapolated signal, e.g. by multiplying it by a half-Hanning window
<xref target="hanning"/>.
</t>
</section>
<section anchor="continuous_chaining" title="Continuous Chaining">
<t>
In some applications, such as Internet radio, it is desirable to cut a long
stream into smaller chains, e.g. so the comment header can be updated.
This can be done simply by separating the input streams into segments and
encoding each segment independently.
The drawback of this approach is that it creates a small discontinuity
at the boundary due to the lossy nature of Opus.
An encoder MAY avoid this discontinuity by using the following procedure:
<list style="numbers">
<t>Encode the last frame of the first segment as an independent frame by
turning off all forms of inter-frame prediction.
De-emphasis is allowed.</t>
<t>Set the granulepos 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>
<figure align="center">
<preamble>
In encoders derived from the reference implementation, inter-frame prediction
can be turned off by calling:
</preamble>
<artwork align="center"><![CDATA[
opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));
]]></artwork>
<postamble>
For best results, this implementation requires that prediction be explicitly
enabled again before resuming normal encoding, even after a reset.
</postamble>
</figure>
</section>
</section>
<section anchor="implementation" title="Implementation Status">
<t>
A brief summary of major implementations of this draft is available
at <eref target="https://wiki.xiph.org/OggOpusImplementation"/>,
along with their status.
</t>
<t>
[Note to RFC Editor: please remove this entire section before
final publication per <xref target="RFC6982"/>.]
</t>
</section>
<section anchor="security" title="Security Considerations">
<t>
Implementations of the Opus codec need to take appropriate security
considerations into account, as outlined in <xref target="RFC4732"/>.
This is just as much a problem for the container as it is for the codec itself.
It is extremely important for the decoder to be robust against malicious
payloads.
Malicious payloads MUST NOT cause the decoder to overrun its allocated memory
or to take an excessive amount of resources to decode.
Although problems in encoders are typically rarer, the same applies to the
encoder.
Malicious audio streams MUST NOT cause the encoder to misbehave because this
would allow an attacker to attack transcoding gateways.
</t>
<t>
Like most other container formats, Ogg Opus streams SHOULD NOT be used with
insecure ciphers or cipher modes that are vulnerable to known-plaintext
attacks.
Elements such as the Ogg page capture pattern and the magic signatures in the
ID header and the comment header all have easily predictable values, in
addition to various elements of the codec data itself.
</t>
</section>
<section anchor="content_type" title="Content Type">
<t>
An "Ogg Opus file" consists of one or more sequentially multiplexed segments,
each containing exactly one Ogg Opus stream.
The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg".
</t>
<figure>
<preamble>
If more specificity is desired, one MAY indicate the presence of Opus streams
using the codecs parameter defined in <xref target="RFC6381"/>, e.g.,
</preamble>
<artwork align="center"><![CDATA[
audio/ogg; codecs=opus
]]></artwork>
<postamble>
for an Ogg Opus file.
</postamble>
</figure>
<t>
The RECOMMENDED filename extension for Ogg Opus files is '.opus'.
</t>
<t>
When Opus is concurrently multiplexed with other streams in an Ogg container,
one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg"
mime-types, as defined in <xref target="RFC5334"/>.
Such streams are not strictly "Ogg Opus files" as described above,
since they contain more than a single Opus stream per sequentially
multiplexed segment, e.g. video or multiple audio tracks.
In such cases the the '.opus' filename extension is NOT RECOMMENDED.
</t>
</section>
<section title="IANA Considerations">
<t>
This document has no actions for IANA.
</t>
</section>
<section anchor="Acknowledgments" title="Acknowledgments">
<t>
Thanks to Greg Maxwell, Christopher "Monty" Montgomery, and Jean-Marc Valin for
their valuable contributions to this document.
Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penqeurc'h for
their feedback based on early implementations.
</t>
</section>
<section title="Copying Conditions">
<t>
The authors agree to grant third parties the irrevocable right to copy, use,
and distribute the work, with or without modification, in any medium, without
royalty, provided that, unless separate permission is granted, redistributed
modified works do not contain misleading author, version, name of work, or
endorsement information.
</t>
</section>
</middle>
<back>
<references title="Normative References">
&rfc2119;
&rfc3533;
&rfc3629;
&rfc5334;
&rfc6381;
&rfc6716;
<reference anchor="EBU-R128" target="https://tech.ebu.ch/loudness">
<front>
<title>Loudness Recommendation EBU R128</title>
<author>
<organization>EBU Technical Committee</organization>
</author>
<date month="August" year="2011"/>
</front>
</reference>
<reference anchor="vorbis-comment"
target="https://www.xiph.org/vorbis/doc/v-comment.html">
<front>
<title>Ogg Vorbis I Format Specification: Comment Field and Header
Specification</title>
<author initials="C." surname="Montgomery"
fullname="Christopher "Monty" Montgomery"/>
<date month="July" year="2002"/>
</front>
</reference>
</references>
<references title="Informative References">
<!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml"?-->
&rfc4732;
&rfc6982;
<reference anchor="flac"
target="https://xiph.org/flac/format.html">
<front>
<title>FLAC - Free Lossless Audio Codec Format Description</title>
<author initials="J." surname="Coalson" fullname="Josh Coalson"/>
<date month="January" year="2008"/>
</front>
</reference>
<reference anchor="hanning"
target="https://en.wikipedia.org/wiki/Hamming_function#Hann_.28Hanning.29_window">
<front>
<title>Hann window</title>
<author>
<organization>Wikipedia</organization>
</author>
<date month="May" year="2013"/>
</front>
</reference>
<reference anchor="linear-prediction"
target="https://en.wikipedia.org/wiki/Linear_predictive_coding">
<front>
<title>Linear Predictive Coding</title>
<author>
<organization>Wikipedia</organization>
</author>
<date month="January" year="2014"/>
</front>
</reference>
<reference anchor="lpc-sample"
target="https://svn.xiph.org/trunk/vorbis/lib/lpc.c">
<front>
<title>Autocorrelation LPC coeff generation algorithm
(Vorbis source code)</title>
<author initials="J." surname="Degener" fullname="Jutta Degener"/>
<author initials="C." surname="Bormann" fullname="Carsten Bormann"/>
<date month="November" year="1994"/>
</front>
</reference>
<reference anchor="replay-gain"
target="https://wiki.xiph.org/VorbisComment#Replay_Gain">
<front>
<title>VorbisComment: Replay Gain</title>
<author initials="C." surname="Parker" fullname="Conrad Parker"/>
<author initials="M." surname="Leese" fullname="Martin Leese"/>
<date month="June" year="2009"/>
</front>
</reference>
<reference anchor="seeking"
target="https://wiki.xiph.org/Seeking">
<front>
<title>Granulepos Encoding and How Seeking Really Works</title>
<author initials="S." surname="Pfeiffer" fullname="Silvia Pfeiffer"/>
<author initials="C." surname="Parker" fullname="Conrad Parker"/>
<author initials="G." surname="Maxwell" fullname="Greg Maxwell"/>
<date month="May" year="2012"/>
</front>
</reference>
<reference anchor="vorbis-mapping"
target="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">
<front>
<title>The Vorbis I Specification, Section 4.3.9 Output Channel Order</title>
<author initials="C." surname="Montgomery"
fullname="Christopher "Monty" Montgomery"/>
<date month="January" year="2010"/>
</front>
</reference>
<reference anchor="vorbis-trim"
target="https://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-130000A.2">
<front>
<title>The Vorbis I Specification, Appendix A: Embedding Vorbis
into an Ogg stream</title>
<author initials="C." surname="Montgomery"
fullname="Christopher "Monty" Montgomery"/>
<date month="November" year="2008"/>
</front>
</reference>
<reference anchor="wave-multichannel"
target="http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx">
<front>
<title>Multiple Channel Audio Data and WAVE Files</title>
<author>
<organization>Microsoft Corporation</organization>
</author>
<date month="March" year="2007"/>
</front>
</reference>
</references>
</back>
</rfc>