Internet DRAFT - draft-terriberry-oggopus
draft-terriberry-oggopus
codec T. Terriberry
Internet-Draft Mozilla Corporation
Intended status: Standards Track R. Lee
Expires: January 17, 2013 Voicetronix
R. Giles
Mozilla Corporation
July 16, 2012
Ogg Encapsulation for the Opus Audio Codec
draft-terriberry-oggopus-01
Abstract
This document defines the Ogg encapsulation for the Opus interactive
speech and audio codec. This allows data encoded in the Opus format
to be stored in an Ogg logical bitstream. 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.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 17, 2013.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Packet Organization . . . . . . . . . . . . . . . . . . . . . 5
4. Granule Position . . . . . . . . . . . . . . . . . . . . . . . 7
4.1. Pre-skip . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.2. PCM Sample Position . . . . . . . . . . . . . . . . . . . 8
4.3. End Trimming . . . . . . . . . . . . . . . . . . . . . . . 9
4.4. Restrictions on the Initial Granule Position . . . . . . . 9
4.5. Seeking and Pre-roll . . . . . . . . . . . . . . . . . . . 10
5. Header Packets . . . . . . . . . . . . . . . . . . . . . . . . 11
5.1. Identification Header . . . . . . . . . . . . . . . . . . 11
5.1.1. Channel Mapping . . . . . . . . . . . . . . . . . . . 15
5.2. Comment Header . . . . . . . . . . . . . . . . . . . . . . 18
6. Packet Size Limits . . . . . . . . . . . . . . . . . . . . . . 22
7. Security Considerations . . . . . . . . . . . . . . . . . . . 23
8. Content Type . . . . . . . . . . . . . . . . . . . . . . . . . 24
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 26
11. Copying Conditions . . . . . . . . . . . . . . . . . . . . . . 27
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
12.1. Normative References . . . . . . . . . . . . . . . . . . . 28
12.2. Informative References . . . . . . . . . . . . . . . . . . 28
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 30
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1. Introduction
The IETF Opus codec is a low-latency audio codec optimized for both
voice and general-purpose audio. See [RFCOpus] for technical
details. This document defines the encapsulation of Opus in a
continuous, logical Ogg bitstream [RFC3533].
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.
This encapsulation defines the required contents of the packet data,
including the necessary headers, the organization of those packets
into a logical stream, and the interpretation of the codec-specific
granule position field. It does not attempt to describe or specify
the existing Ogg container format. Readers unfamiliar with the basic
concepts mentioned above are encouraged to review the details in
[RFC3533].
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2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
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".
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3. Packet Organization
An Opus stream is organized as follows.
There are two mandatory header packets. The granule position of the
pages on which these packets complete MUST be zero.
The first packet in the logical Ogg bitstream MUST contain the
identification (ID) header, which uniquely identifies a stream as
Opus audio. The format of this header is defined in Section 5.1. It
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.
The second packet in the logical Ogg bitstream MUST contain the
comment header, which contains user-supplied metadata. The format of
this header is defined in Section 5.2. It MAY span 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.
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, e.g.,
multichannel audio. The value N is specified in the ID header (see
Section 5.1.1), and is fixed over the entire length of the logical
Ogg bitstream.
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
[RFCOpus]. The remaining Opus packet is packed at the end of the Ogg
packet using the regular, undelimited framing from Section 3 of
[RFCOpus]. All of the Opus packets in a single Ogg packet MUST be
constrained to have the same duration. The duration and coding modes
of each Opus packet are contained in the TOC (table of contents)
sequence in the first few bytes. 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 an Opus packet with an illegal TOC
sequence.
The first audio data page SHOULD NOT have the 'continued packet' flag
set (which would indicated 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 an Opus packet with an illegal TOC sequence. The last
page SHOULD have the 'end of stream' flag set, but implementations
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should 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'.
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4. Granule Position
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.
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.
The duration of an Opus packet may 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.
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. In order to support capturing a stream that uses
discontinuous transmission (DTX), an encoder SHOULD emit packets that
explicitly request the use of Packet Loss Concealment (PLC) (i.e.,
with a frame length of 0, as defined in Section 3.2.1 of [RFCOpus])
in place of the packets that were not transmitted.
4.1. Pre-skip
There is some amount of latency introduced during the decoding
process, to allow for overlap in the MDCT modes, stereo mixing in the
LP modes, and resampling, and the encoder will introduce even more
latency (though the exact amount is not specified). Therefore, the
first few samples produced by the decoder do not correspond to real
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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.
A 'pre-skip' field in the ID header (see Section 5.1) signals the
number of samples which should be skipped (decoded but discarded) at
the beginning of the stream. This provides sufficient history to the
decoder so that it has already converged before the stream's output
begins. It may also be used to perform sample-accurate cropping of
existing encoded streams. 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.
4.2. PCM Sample Position
The PCM sample position is determined from the granule position using
the formula
'PCM sample position' = 'granule position' - 'pre-skip' .
For example, if the granule position of the first audio data page is
59,971, and the pre-skip is 11,971, then the PCM sample position of
the last decoded sample from that page is 48,000. This can be
converted into a playback time using the formula
'PCM sample position'
'playback time' = --------------------- .
48000.0
The initial PCM sample position before any samples are played is
normally '0'. In this case, the PCM sample position of the first
audio sample to be played starts at '1', because it marks the time on
the clock _after_ that sample has been played, and a stream that is
exactly one second long has a final PCM sample position of '48000',
as in the example here.
Vorbis streams use a granule position smaller than the number of
audio samples contained in the first audio data page to indicate that
some of those samples must be trimmed from the output (see
[vorbis-trim]). However, to do so, Vorbis requires that the first
audio data page contains exactly two packets, in order to allow the
decoder to perform PCM position adjustments before needing to return
any PCM data. Opus uses the pre-skip mechanism for this purpose
instead, since the encoder may introduce more than a single packet's
worth of latency, and since very large packets in streams with a very
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large number of channels might not fit on a single page.
4.3. End Trimming
The page with the 'end of stream' flag set MAY have a granule
position that indicates the page contains less audio data than would
normally be returned by decoding up through the final packet. This
is used to end the stream somewhere other than an even frame
boundary. The granule position of the most recent audio data page
with completed packets is used to make this determination, or '0' is
used if there were no previous audio data pages with a completed
packet. The difference between these granule positions indicates how
many samples to keep after decoding the packets that completed on the
final page. The remaining samples are discarded. The number of
discarded samples SHOULD be no larger than the number decoded from
the last packet.
4.4. Restrictions on the Initial Granule Position
The granule position of the first audio data page with a completed
packet MAY be larger than the number of samples contained in packets
that complete on that page, however it MUST NOT be smaller, unless
that page has the 'end of stream' flag set. Allowing a granule
position larger than the number of samples allows the beginning of a
stream to be cropped or a live stream to be joined without rewriting
the granule position of all the remaining pages. This means that the
PCM sample position just before the first sample to be played may be
larger than '0'. Synchronization when multiplexing with other
logical streams still uses the PCM sample position relative to '0' to
compute sample times. This does not affect the behavior of pre-skip:
exactly 'pre-skip' samples should be skipped from the beginning of
the decoded output, even if the initial PCM sample position is
greater than zero.
On the other hand, a granule position that is smaller than the number
of decoded samples prevents a demuxer from working backwards to
assign each packet or each individual sample a valid granule
position, since granule positions must be 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. If that
page has the 'end of stream' flag set, a demuxer can work forwards
from the granule position '0', but MUST reject as invalid any stream
where the granule position is smaller than the 'pre-skip' amount.
This would indicate that more samples should be skipped from the
initial decoded output than exist in the stream.
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4.5. Seeking and Pre-roll
Seeking in Ogg files is best performed using a bisection search for a
page whose granule position corresponds to a PCM position at or
before the seek target. With appropriately weighted bisection,
accurate seeking can be performed with just three or four bisections
even in multi-gigabyte files. See [seeking] for general
implementation guidance.
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.
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5. Header Packets
An Opus stream contains exactly two mandatory header packets.
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 1: ID Header Packet
The fields in the identification (ID) header have the following
meaning:
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'
0x48 'H'
0x65 'e'
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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, 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.
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 constructing cropped Ogg Opus streams, a
pre-skip of at least 3,840 samples (80 ms) is RECOMMENDED to
ensure complete convergence.
5. *Input Sample Rate* (32 bits, unsigned, little endian):
This field is _not_ the sample rate to use for playback of the
encoded data.
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Opus has a handful of coding modes, with internal audio
bandwidths of 4, 6, 8, 12, and 20 kHz. Each packet in the stream
may 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.
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 highest supported rate above
this and resample.
4. Otherwise, decode at 48 kHz and resample.
However, the 'Input Sample Rate' field allows the encoder to pass
the sample rate of the original input stream as metadata. This
may be 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.
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).
6. *Output Gain* (16 bits, signed, little endian):
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). To apply
the gain, a decoder could use
sample *= pow(10, output_gain/(20.0*256)) ,
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where output_gain is the raw 16-bit value from the header.
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 Section 5.2)
or a user-facing volume knob, the adjustment MUST be applied in
addition to this output gain in order to achieve playback at the
desired volume.
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. The output gain should only be
nonzero when the gain is adjusted after encoding, or when the
user wishes 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_TRACK_GAIN (see below) without saturating.
7. *Channel Mapping Family* (8 bits, unsigned):
This octet indicates the order and semantic meaning of the
various channels encoded in each Ogg packet.
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 Section 5.1.1.
8. *Channel Mapping Table*: 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 Section 5.1.1.
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
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the ID header does not complete 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 these channels
is 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 2.
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 2: 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 required 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.
2. *Coupled Stream Count* 'M' (8 bits, unsigned): This is the number
of streams whose decoders should be configured to produce two
channels. 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 contents being
encoded. The original channel count of the encoder input is not
preserved by the lossy compression.
Regardless of the internal channel count, any Opus stream can be
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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 in stereo mode, and the remaining N-M decoders are to
be initialized in mono mode. 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 should 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, 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.
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.
After producing the output channels, the channel mapping family
determines the semantic meaning of each one. Currently there are
three defined mapping families, although more may be added:
o Family 0 (RTP mapping):
Allowed numbers of channels: 1 or 2.
* 1 channel: monophonic (mono).
* 2 channels: stereo (left, right).
*Special mapping*: This channel mapping value also indicates that
the contents consists of a single Opus stream that is stereo if
and only if C==2, with stream index 0 mapped to channel 0, and (if
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stereo) stream index 1 mapped to channel 1. When the 'channel
mapping family' octet has this value, the channel mapping table
MUST be omitted from the ID header packet.
o Family 1 (Vorbis channel order):
Allowed numbers of channels: 1...8.
Channel meanings depend on the number of channels. See
[vorbis-mapping] for the assignments from output channel number to
specific speaker locations.
o Family 255 (no defined channel meaning):
Allowed numbers of channels: 1...255.
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.
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.
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.
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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 3: Comment Header Packet
The comment header consists of a 64-bit magic signature, followed by
data in the same format as the [vorbis-comment] header used in Ogg
Vorbis (without the final "framing bit"), Ogg Theora, and Speex.
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'
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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 NUL octet is
required.
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.
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.
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6. *User Comment #i String* (variable length, UTF-8 vector):
This field contains a single user comment string. There is one
for each user comment indicated by the 'user comment list length'
field.
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 may help 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.
The user comment strings follow the NAME=value format described by
[vorbis-comment] with the same recommended tag names. One new
comment tag is introduced for Ogg Opus:
R128_TRACK_GAIN=-573
representing the volume shift needed to normalize the track's volume.
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.
An Ogg Opus file MUST NOT have more than one such tag, and if present
its value MUST be an integer from -32768 to 32767, inclusive,
represented in ASCII with no whitespace. If present, it 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, it MUST be applied _in
addition_ to the 'output gain' value. If an encoder wishes to use
R128 normalization, and the output gain is not otherwise constrained
or specified, the encoder SHOULD write the R128 gain into the 'output
gain' field and store a tag containing "R128_TRACK_GAIN=0". That is,
it should assume that by default tools will respect the 'output gain'
field, and not the comment tag. If a tool modifies the ID header's
'output gain' field, it MUST also update or remove the
R128_TRACK_GAIN comment tag.
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.
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There is no Opus comment tag corresponding to REPLAYGAIN_ALBUM_GAIN.
That information should instead be stored in the ID header's 'output
gain' field.
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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. Encoders SHOULD use no more padding
than required 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.
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, which are required to run
in the MDCT mode, 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 LP 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 MDCT-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.
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7. 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.
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.
Like most other container formats, Ogg Opus files 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.
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8. 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". 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].
If more specificity is desired, one MAY indicate the presence of Opus
streams using the codecs parameter defined in [RFC6381], e.g.,
audio/ogg; codecs=opus
for an Ogg Opus file.
The RECOMMENDED filename extension for Ogg Opus files is '.opus'.
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9. IANA Considerations
This document has no actions for IANA.
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10. Acknowledgments
Thanks to Ralph Giles, Greg Maxwell, Christopher "Monty" Montgomery,
and Jean-Marc Valin for their valuable contributions to this
document. Additional thanks to Andrew D'Addesio, Ralph Giles, Greg
Maxwell, and Vincent Penqeurc'h for their feedback based on early
implementations.
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11. Copying Conditions
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.
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12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, November 2003.
[RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
RFC 3533, May 2003.
[RFC5334] Goncalves, I., Pfeiffer, S., and C. Montgomery, "Ogg Media
Types", RFC 5334, September 2008.
[RFC6381] Gellens, R., Singer, D., and P. Frojdh, "The 'Codecs' and
'Profiles' Parameters for "Bucket" Media Types", RFC 6381,
August 2011.
[RFCOpus] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC XXXX.
[EBU-R128]
""Loudness Recommendation EBU R128",
<http://tech.ebu.ch/loudness>.
[vorbis-comment]
Montgomery, C., "Ogg Vorbis I Format Specification:
Comment Field and Header Specification",
<http://www.xiph.org/vorbis/doc/v-comment.html>.
[vorbis-mapping]
Montgomery, C., "The Vorbis I Specification, Section 4.3.9
Output Channel Order", <http://www.xiph.org/vorbis/doc/
Vorbis_I_spec.html#x1-800004.3.9>.
12.2. Informative References
[RFC4732] Handley, M., Rescorla, E., and IAB, "Internet Denial-of-
Service Considerations", RFC 4732, December 2006.
[replay-gain]
Parker, C. and M. Leese, "VorbisComment: Replay Gain",
<http://wiki.xiph.org/VorbisComment#Replay_Gain>.
[seeking] Pfeiffer, S., Parker, C., and G. Maxwell, "Granulepos
Encoding and How Seeking Really Works",
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<http://wiki.xiph.org/Seeking>.
[vorbis-trim]
Montgomery, C., "The Vorbis I Specification, Appendix A
Embedding Vorbis into an Ogg stream", <http://xiph.org/
vorbis/doc/Vorbis_I_spec.html#x1-130000A.2>.
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Authors' Addresses
Timothy B. Terriberry
Mozilla Corporation
650 Castro Street
Mountain View, CA 94041
USA
Phone: +1 650 903-0800
Email: tterribe@xiph.org
Ron Lee
Voicetronix
246 Pulteney Street, Level 1
Adelaide, SA 5000
Australia
Phone: +61 8 8232 9112
Email: ron@debian.org
Ralph Giles
Mozilla Corporation
163 West Hastings Street
Vancouver, BC V6B 1H5
Canada
Phone: +1 604 778 1540
Email: giles@xiph.org
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