| Rfc | 7845 |
|---|
| Title | Ogg Encapsulation for the Opus Audio Codec |
|---|
| Author | T. Terriberry, R. Lee,
R. Giles |
|---|
| Date | April 2016 |
|---|
| Format: | TXT, HTML |
|---|
| Updates | RFC5334 |
|---|
| Status: | PROPOSED STANDARD |
|---|
|
Internet Engineering Task Force (IETF) T. Terriberry
Request for Comments: 7845 Mozilla Corporation
Updates: 5334 R. Lee
Category: Standards Track Voicetronix
ISSN: 2070-1721 R. Giles
Mozilla Corporation
April 2016
Ogg Encapsulation for the Opus Audio Codec
Abstract
This document defines the Ogg encapsulation for the Opus interactive
speech and audio codec. This allows data encoded in the Opus format
to be stored in an Ogg logical bitstream.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7845.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Packet Organization . . . . . . . . . . . . . . . . . . . . . 4
4. Granule Position . . . . . . . . . . . . . . . . . . . . . . 6
4.1. Repairing Gaps in Real-Time Streams . . . . . . . . . . . 7
4.2. Pre-skip . . . . . . . . . . . . . . . . . . . . . . . . 9
4.3. PCM Sample Position . . . . . . . . . . . . . . . . . . . 9
4.4. End Trimming . . . . . . . . . . . . . . . . . . . . . . 10
4.5. Restrictions on the Initial Granule Position . . . . . . 10
4.6. Seeking and Pre-roll . . . . . . . . . . . . . . . . . . 11
5. Header Packets . . . . . . . . . . . . . . . . . . . . . . . 12
5.1. Identification Header . . . . . . . . . . . . . . . . . . 12
5.1.1. Channel Mapping . . . . . . . . . . . . . . . . . . . 16
5.2. Comment Header . . . . . . . . . . . . . . . . . . . . . 22
5.2.1. Tag Definitions . . . . . . . . . . . . . . . . . . . 25
6. Packet Size Limits . . . . . . . . . . . . . . . . . . . . . 26
7. Encoder Guidelines . . . . . . . . . . . . . . . . . . . . . 27
7.1. LPC Extrapolation . . . . . . . . . . . . . . . . . . . . 28
7.2. Continuous Chaining . . . . . . . . . . . . . . . . . . . 28
8. Security Considerations . . . . . . . . . . . . . . . . . . . 29
9. Content Type . . . . . . . . . . . . . . . . . . . . . . . . 30
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 32
11.1. Normative References . . . . . . . . . . . . . . . . . . 32
11.2. Informative References . . . . . . . . . . . . . . . . . 33
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 34
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35
1. Introduction
The IETF Opus codec is a low-latency audio codec optimized for both
voice and general-purpose audio. See [RFC6716] for technical
details. This document defines the encapsulation of Opus in a
continuous, logical Ogg bitstream [RFC3533]. Ogg encapsulation
provides Opus with a long-term storage format supporting all of the
essential features, including metadata, fast and accurate seeking,
corruption detection, recapture after errors, low overhead, and the
ability to multiplex Opus with other codecs (including video) with
minimal buffering. It also provides a live streamable format capable
of delivery over a reliable stream-oriented transport, without
requiring all the data (or even the total length of the data)
up-front, in a form that is identical to the on-disk storage format.
Ogg bitstreams are made up of a series of "pages", each of which
contains data from one or more "packets". Pages are the fundamental
unit of multiplexing in an Ogg stream. Each page is associated with
a particular logical stream and contains a capture pattern and
checksum, flags to mark the beginning and end of the logical stream,
and a "granule position" that represents an absolute position in the
stream, to aid seeking. A single page can contain up to 65,025
octets of packet data from up to 255 different packets. Packets can
be split arbitrarily across pages and continued from one page to the
next (allowing packets much larger than would fit on a single page).
Each page contains "lacing values" that indicate how the data is
partitioned into packets, allowing a demultiplexer (demuxer) to
recover the packet boundaries without examining the encoded data. A
packet is said to "complete" on a page when the page contains the
final lacing value corresponding to that packet.
This encapsulation defines the contents of the packet data, including
the necessary headers, the organization of those packets into a
logical stream, and the interpretation of the codec-specific granule
position field. It does not attempt to describe or specify the
existing Ogg container format. Readers unfamiliar with the basic
concepts mentioned above are encouraged to review the details in
[RFC3533].
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
[RFC2119].
3. Packet Organization
An Ogg Opus stream is organized as follows (see Figure 1 for an
example).
Page 0 Pages 1 ... n Pages (n+1) ...
+------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
| | | | | | | | | | | | |
|+----------+| |+-----------------+| |+-------------------+ +-----
|||ID Header|| || Comment Header || ||Audio Data Packet 1| | ...
|+----------+| |+-----------------+| |+-------------------+ +-----
| | | | | | | | | | | | |
+------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +--
^ ^ ^
| | |
| | Mandatory Page Break
| |
| ID header is contained on a single page
|
'Beginning Of Stream'
Figure 1: Example Packet Organization for a Logical Ogg Opus Stream
There are two mandatory header packets. The first packet in the
logical Ogg bitstream MUST contain the identification (ID) header,
which uniquely identifies a stream as Opus audio. The format of this
header is defined in Section 5.1. It is placed alone (without any
other packet data) on the first page of the logical Ogg bitstream and
completes on that page. This page has its 'beginning of stream' flag
set.
The second packet in the logical Ogg bitstream MUST contain the
comment header, which contains user-supplied metadata. The format of
this header is defined in Section 5.2. It MAY span multiple pages,
beginning on the second page of the logical stream. However many
pages it spans, the comment header packet MUST finish the page on
which it completes.
All subsequent pages are audio data pages, and the Ogg packets they
contain are audio data packets. Each audio data packet contains one
Opus packet for each of N different streams, where N is typically one
for mono or stereo, but MAY be greater than one for multichannel
audio. The value N is specified in the ID header (see
Section 5.1.1), and is fixed over the entire length of the logical
Ogg bitstream.
The first (N - 1) Opus packets, if any, are packed one after another
into the Ogg packet, using the self-delimiting framing from
Appendix B of [RFC6716]. The remaining Opus packet is packed at the
end of the Ogg packet using the regular, undelimited framing from
Section 3 of [RFC6716]. All of the Opus packets in a single Ogg
packet MUST be constrained to have the same duration. An
implementation of this specification SHOULD treat any Opus packet
whose duration is different from that of the first Opus packet in an
Ogg packet as if it were a malformed Opus packet with an invalid
Table Of Contents (TOC) sequence.
The TOC sequence at the beginning of each Opus packet indicates the
coding mode, audio bandwidth, channel count, duration (frame size),
and number of frames per packet, as described in Section 3.1
of [RFC6716]. The coding mode is one of SILK, Hybrid, or Constrained
Energy Lapped Transform (CELT). The combination of coding mode,
audio bandwidth, and frame size is referred to as the configuration
of an Opus packet.
Packets are placed into Ogg pages in order until the end of stream.
Audio data packets might span page boundaries. The first audio data
page could have the 'continued packet' flag set (indicating the first
audio data packet is continued from a previous page) if, for example,
it was a live stream joined mid-broadcast, with the headers pasted on
the front. If a page has the 'continued packet' flag set and one of
the following conditions is also true:
o the previous page with packet data does not end in a continued
packet (does not end with a lacing value of 255) OR
o the page sequence numbers are not consecutive,
then a demuxer MUST NOT attempt to decode the data for the first
packet on the page unless the demuxer has some special knowledge that
would allow it to interpret this data despite the missing pieces. An
implementation MUST treat a zero-octet audio data packet as if it
were a malformed Opus packet as described in Section 3.4
of [RFC6716].
A logical stream ends with a page with the 'end of stream' flag set,
but implementations need to be prepared to deal with truncated
streams that do not have a page marked 'end of stream'. There is no
reason for the final packet on the last page to be a continued
packet, i.e., for the final lacing value to be 255. However,
demuxers might encounter such streams, possibly as the result of a
transfer that did not complete or of corruption. If a packet
continues onto a subsequent page (i.e., when the page ends with a
lacing value of 255) and one of the following conditions is also
true:
o the next page with packet data does not have the 'continued
packet' flag set, OR
o there is no next page with packet data, OR
o the page sequence numbers are not consecutive,
then a demuxer MUST NOT attempt to decode the data from that packet
unless the demuxer has some special knowledge that would allow it to
interpret this data despite the missing pieces. There MUST NOT be
any more pages in an Opus logical bitstream after a page marked 'end
of stream'.
4. Granule Position
The granule position MUST be zero for the ID header page and the page
where the comment header completes. That is, the first page in the
logical stream and the last header page before the first audio data
page both have a granule position of zero.
The granule position of an audio data page encodes the total number
of PCM samples in the stream up to and including the last fully
decodable sample from the last packet completed on that page. The
granule position of the first audio data page will usually be larger
than zero, as described in Section 4.5.
A page that is entirely spanned by a single packet (that completes on
a subsequent page) has no granule position, and the granule position
field is set to the special value '-1' in two's complement.
The granule position of an audio data page is in units of PCM audio
samples at a fixed rate of 48 kHz (per channel; a stereo stream's
granule position does not increment at twice the speed of a mono
stream). It is possible to run an Opus decoder at other sampling
rates, but all Opus packets encode samples at a sampling rate that
evenly divides 48 kHz. Therefore, the value in the granule position
field always counts samples assuming a 48 kHz decoding rate, and the
rest of this specification makes the same assumption.
The duration of an Opus packet as defined in [RFC6716] can be any
multiple of 2.5 ms, up to a maximum of 120 ms. This duration is
encoded in the TOC sequence at the beginning of each packet. The
number of samples returned by a decoder corresponds to this duration
exactly, even for the first few packets. For example, a 20 ms packet
fed to a decoder running at 48 kHz will always return 960 samples. A
demuxer can parse the TOC sequence at the beginning of each Ogg
packet to work backwards or forwards from a packet with a known
granule position (i.e., the last packet completed on some page) in
order to assign granule positions to every packet, or even every
individual sample. The one exception is the last page in the stream,
as described below.
All other pages with completed packets after the first MUST have a
granule position equal to the number of samples contained in packets
that complete on that page plus the granule position of the most
recent page with completed packets. This guarantees that a demuxer
can assign individual packets the same granule position when working
forwards as when working backwards. For this to work, there cannot
be any gaps.
4.1. Repairing Gaps in Real-Time Streams
In order to support capturing a real-time stream that has lost or not
transmitted packets, a multiplexer (muxer) SHOULD emit packets that
explicitly request the use of Packet Loss Concealment (PLC) in place
of the missing packets. Implementations that fail to do so still
MUST NOT increment the granule position for a page by anything other
than the number of samples contained in packets that actually
complete on that page.
Only gaps that are a multiple of 2.5 ms are repairable, as these are
the only durations that can be created by packet loss or
discontinuous transmission. Muxers need not handle other gap sizes.
Creating the necessary packets involves synthesizing a TOC byte
(defined in Section 3.1 of [RFC6716]) -- and whatever additional
internal framing is needed -- to indicate the packet duration for
each stream. The actual length of each missing Opus frame inside the
packet is zero bytes, as defined in Section 3.2.1 of [RFC6716].
Zero-byte frames MAY be packed into packets using any of codes 0, 1,
2, or 3. When successive frames have the same configuration, the
higher code packings reduce overhead. Likewise, if the TOC
configuration matches, the muxer MAY further combine the empty frames
with previous or subsequent nonzero-length frames (using code 2 or
variable bitrate (VBR) code 3).
[RFC6716] does not impose any requirements on the PLC, but this
section outlines choices that are expected to have a positive
influence on most PLC implementations, including the reference
implementation. Synthesized TOC sequences SHOULD maintain the same
mode, audio bandwidth, channel count, and frame size as the previous
packet (if any). This is the simplest and usually the most well-
tested case for the PLC to handle and it covers all losses that do
not include a configuration switch, as defined in Section 4.5
of [RFC6716].
When a previous packet is available, keeping the audio bandwidth and
channel count the same allows the PLC to provide maximum continuity
in the concealment data it generates. However, if the size of the
gap is not a multiple of the most recent frame size, then the frame
size will have to change for at least some frames. Such changes
SHOULD be delayed as long as possible to simplify things for PLC
implementations.
As an example, a 95 ms gap could be encoded as nineteen 5 ms frames
in two bytes with a single constant bitrate (CBR) code 3 packet. If
the previous frame size was 20 ms, using four 20 ms frames followed
by three 5 ms frames requires 4 bytes (plus an extra byte of Ogg
lacing overhead), but allows the PLC to use its well-tested steady
state behavior for as long as possible. The total bitrate of the
latter approach, including Ogg overhead, is about 0.4 kbps, so the
impact on file size is minimal.
Changing modes is discouraged, since this causes some decoder
implementations to reset their PLC state. However, SILK and Hybrid
mode frames cannot fill gaps that are not a multiple of 10 ms. If
switching to CELT mode is needed to match the gap size, a muxer
SHOULD do so at the end of the gap to allow the PLC to function for
as long as possible.
In the example above, if the previous frame was a 20 ms SILK mode
frame, the better solution is to synthesize a packet describing four
20 ms SILK frames, followed by a packet with a single 10 ms SILK
frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms
gap. This also requires four bytes to describe the synthesized
packet data (two bytes for a CBR code 3 and one byte each for two
code 0 packets) but three bytes of Ogg lacing overhead are needed to
mark the packet boundaries. At 0.6 kbps, this is still a minimal
bitrate impact over a naive, low-quality solution.
Since medium-band audio is an option only in the SILK mode, wideband
frames SHOULD be generated if switching from that configuration to
CELT mode, to ensure that any PLC implementation that does try to
migrate state between the modes will be able to preserve all of the
available audio bandwidth.
4.2. Pre-skip
There is some amount of latency introduced during the decoding
process, to allow for overlap in the CELT mode, stereo mixing in the
SILK mode, and resampling. The encoder might have introduced
additional latency through its own resampling and analysis (though
the exact amount is not specified). Therefore, the first few samples
produced by the decoder do not correspond to real input audio, but
are instead composed of padding inserted by the encoder to compensate
for this latency. These samples need to be stored and decoded, as
Opus is an asymptotically convergent predictive codec, meaning the
decoded contents of each frame depend on the recent history of
decoder inputs. However, a player will want to skip these samples
after decoding them.
A 'pre-skip' field in the ID header (see Section 5.1) signals the
number of samples that SHOULD be skipped (decoded but discarded) at
the beginning of the stream, though some specific applications might
have a reason for looking at that data. This amount need not be a
multiple of 2.5 ms, MAY be smaller than a single packet, or MAY span
the contents of several packets. These samples are not valid audio.
For example, if the first Opus frame uses the CELT mode, it will
always produce 120 samples of windowed overlap-add data. However,
the overlap data is initially all zeros (since there is no prior
frame), meaning this cannot, in general, accurately represent the
original audio. The SILK mode requires additional delay to account
for its analysis and resampling latency. The encoder delays the
original audio to avoid this problem.
The 'pre-skip' field MAY also be used to perform sample-accurate
cropping of already encoded streams. In this case, a value of at
least 3840 samples (80 ms) provides sufficient history to the decoder
that it will have converged before the stream's output begins.
4.3. PCM Sample Position
The PCM sample position is determined from the granule position using
the following formula:
'PCM sample position' = 'granule position' - 'pre-skip'
For example, if the granule position of the first audio data page is
59,971, and the pre-skip is 11,971, then the PCM sample position of
the last decoded sample from that page is 48,000.
This can be converted into a playback time using the following
formula:
'PCM sample position'
'playback time' = ---------------------
48000.0
The initial PCM sample position before any samples are played is
normally '0'. In this case, the PCM sample position of the first
audio sample to be played starts at '1', because it marks the time on
the clock _after_ that sample has been played, and a stream that is
exactly one second long has a final PCM sample position of '48000',
as in the example here.
Vorbis streams use a granule position smaller than the number of
audio samples contained in the first audio data page to indicate that
some of those samples are trimmed from the output (see
[VORBIS-TRIM]). However, to do so, Vorbis requires that the first
audio data page contains exactly two packets, in order to allow the
decoder to perform PCM position adjustments before needing to return
any PCM data. Opus uses the pre-skip mechanism for this purpose
instead, since the encoder might introduce more than a single
packet's worth of latency, and since very large packets in streams
with a very large number of channels might not fit on a single page.
4.4. End Trimming
The page with the 'end of stream' flag set MAY have a granule
position that indicates the page contains less audio data than would
normally be returned by decoding up through the final packet. This
is used to end the stream somewhere other than an even frame
boundary. The granule position of the most recent audio data page
with completed packets is used to make this determination, or '0' is
used if there were no previous audio data pages with a completed
packet. The difference between these granule positions indicates how
many samples to keep after decoding the packets that completed on the
final page. The remaining samples are discarded. The number of
discarded samples SHOULD be no larger than the number decoded from
the last packet.
4.5. Restrictions on the Initial Granule Position
The granule position of the first audio data page with a completed
packet MAY be larger than the number of samples contained in packets
that complete on that page. However, it MUST NOT be smaller, unless
that page has the 'end of stream' flag set. Allowing a granule
position larger than the number of samples allows the beginning of a
stream to be cropped or a live stream to be joined without rewriting
the granule position of all the remaining pages. This means that the
PCM sample position just before the first sample to be played MAY be
larger than '0'. Synchronization when multiplexing with other
logical streams still uses the PCM sample position relative to '0' to
compute sample times. This does not affect the behavior of pre-skip:
exactly 'pre-skip' samples SHOULD be skipped from the beginning of
the decoded output, even if the initial PCM sample position is
greater than zero.
On the other hand, a granule position that is smaller than the number
of decoded samples prevents a demuxer from working backwards to
assign each packet or each individual sample a valid granule
position, since granule positions are non-negative. An
implementation MUST treat any stream as invalid if the granule
position is smaller than the number of samples contained in packets
that complete on the first audio data page with a completed packet,
unless that page has the 'end of stream' flag set. It MAY defer this
action until it decodes the last packet completed on that page.
If that page has the 'end of stream' flag set, a demuxer MUST treat
any stream as invalid if its granule position is smaller than the
'pre-skip' amount. This would indicate that there are more samples
to be skipped from the initial decoded output than exist in the
stream. If the granule position is smaller than the number of
decoded samples produced by the packets that complete on that page,
then a demuxer MUST use an initial granule position of '0', and can
work forwards from '0' to timestamp individual packets. If the
granule position is larger than the number of decoded samples
available, then the demuxer MUST still work backwards as described
above, even if the 'end of stream' flag is set, to determine the
initial granule position, and thus the initial PCM sample position.
Both of these will be greater than '0' in this case.
4.6. Seeking and Pre-roll
Seeking in Ogg files is best performed using a bisection search for a
page whose granule position corresponds to a PCM position at or
before the seek target. With appropriately weighted bisection,
accurate seeking can be performed in just one or two bisections on
average, even in multi-gigabyte files. See [SEEKING] for an example
of general implementation guidance.
When seeking within an Ogg Opus stream, an implementation SHOULD
start decoding (and discarding the output) at least 3840 samples
(80 ms) prior to the seek target in order to ensure that the output
audio is correct by the time it reaches the seek target. This
"pre-roll" is separate from, and unrelated to, the pre-skip used at
the beginning of the stream. If the point 80 ms prior to the seek
target comes before the initial PCM sample position, an
implementation SHOULD start decoding from the beginning of the
stream, applying pre-skip as normal, regardless of whether the pre-
skip is larger or smaller than 80 ms, and then continue to discard
samples to reach the seek target (if any).
5. Header Packets
An Ogg Opus logical stream contains exactly two mandatory header
packets: an identification header and a comment header.
5.1. Identification Header
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'O' | 'p' | 'u' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'H' | 'e' | 'a' | 'd' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Version = 1 | Channel Count | Pre-skip |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Input Sample Rate (Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Output Gain (Q7.8 in dB) | Mapping Family| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ :
| |
: Optional Channel Mapping Table... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: ID Header Packet
The fields in the identification (ID) header have the following
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'
0x61 'a'
0x64 'd'
Starting with "Op" helps distinguish it from audio data packets,
as this is an invalid TOC sequence.
2. Version (8 bits, unsigned):
The version number MUST always be '1' for this version of the
encapsulation specification. Implementations SHOULD treat
streams where the upper four bits of the version number match
that of a recognized specification as backwards compatible with
that specification. That is, the version number can be split
into "major" and "minor" version sub-fields, with changes to the
minor sub-field (in the lower four bits) signaling compatible
changes. For example, an implementation of this specification
SHOULD accept any stream with a version number of '15' or less,
and SHOULD assume any stream with a version number '16' or
greater is incompatible. The initial version '1' was chosen to
keep implementations from relying on this octet as a null
terminator for the "OpusHead" string.
3. Output Channel Count 'C' (8 bits, unsigned):
This is the number of output channels. This might be different
than the number of encoded channels, which can change on a
packet-by-packet basis. This value MUST NOT be zero. The
maximum allowable value depends on the channel mapping family,
and might be as large as 255. See Section 5.1.1 for details.
4. Pre-skip (16 bits, unsigned, little endian):
This is the number of samples (at 48 kHz) to discard from the
decoder output when starting playback, and also the number to
subtract from a page's granule position to calculate its PCM
sample position. When cropping the beginning of existing Ogg
Opus streams, a pre-skip of at least 3,840 samples (80 ms) is
RECOMMENDED to ensure complete convergence in the decoder.
5. Input Sample Rate (32 bits, unsigned, little endian):
This is the sample rate of the original input (before encoding),
in Hz. This field is _not_ the sample rate to use for playback
of the encoded data.
Opus can switch between internal audio bandwidths of 4, 6, 8, 12,
and 20 kHz. Each packet in the stream can have a different audio
bandwidth. Regardless of the audio bandwidth, the reference
decoder supports decoding any stream at a sample rate of 8, 12,
16, 24, or 48 kHz. The original sample rate of the audio passed
to the encoder is not preserved by the lossy compression.
An Ogg Opus player SHOULD select the playback sample rate
according to the following procedure:
1. If the hardware supports 48 kHz playback, decode at 48 kHz.
2. Otherwise, if the hardware's highest available sample rate is
a supported rate, decode at this sample rate.
3. Otherwise, if the hardware's highest available sample rate is
less than 48 kHz, decode at the next higher Opus supported
rate above the highest available hardware rate and resample.
4. Otherwise, decode at 48 kHz and resample.
However, the 'input sample rate' field allows the muxer to pass
the sample rate of the original input stream as metadata. This
is useful when the user requires the output sample rate to match
the input sample rate. For example, when not playing the output,
an implementation writing PCM format samples to disk might choose
to resample the audio back to the original input sample rate to
reduce surprise to the user, who might reasonably expect to get
back a file with the same sample rate.
A value of zero indicates "unspecified". Muxers SHOULD write the
actual input sample rate or zero, but implementations that do
something with this field SHOULD take care to behave sanely if
given crazy values (e.g., do not actually upsample the output to
10 MHz if requested). Implementations SHOULD support input
sample rates between 8 kHz and 192 kHz (inclusive). Rates
outside this range MAY be ignored by falling back to the default
rate of 48 kHz instead.
6. Output Gain (16 bits, signed, little endian):
This is a gain to be applied when decoding. It is 20*log10 of
the factor by which to scale the decoder output to achieve the
desired playback volume, stored in a 16-bit, signed, two's
complement fixed-point value with 8 fractional bits (i.e.,
Q7.8 [Q-NOTATION]).
To apply the gain, an implementation could use the following:
sample *= pow(10, output_gain/(20.0*256))
where 'output_gain' is the raw 16-bit value from the header.
Players and media frameworks SHOULD apply it by default. If a
player chooses to apply any volume adjustment or gain
modification, such as the R128_TRACK_GAIN (see Section 5.2), the
adjustment MUST be applied in addition to this output gain in
order to achieve playback at the normalized volume.
A muxer SHOULD set this field to zero, and instead apply any gain
prior to encoding, when this is possible and does not conflict
with the user's wishes. A nonzero output gain indicates the gain
was adjusted after encoding, or that a user wished to adjust the
gain for playback while preserving the ability to recover the
original signal amplitude.
Although the output gain has enormous range (+/- 128 dB, enough
to amplify inaudible sounds to the threshold of physical pain),
most applications can only reasonably use a small portion of this
range around zero. The large range serves in part to ensure that
gain can always be losslessly transferred between OpusHead and
R128 gain tags (see below) without saturating.
7. Channel Mapping Family (8 bits, unsigned):
This octet indicates the order and semantic meaning of the output
channels.
Each currently specified value of this octet indicates a mapping
family, which defines a set of allowed channel counts, and the
ordered set of channel names for each allowed channel count. The
details are described in Section 5.1.1.
8. Channel Mapping Table:
This table defines the mapping from encoded streams to output
channels. Its contents are specified in Section 5.1.1.
All fields in the ID headers are REQUIRED, except for 'channel
mapping table', which MUST be omitted when the channel mapping family
is 0, but is REQUIRED otherwise. Implementations SHOULD treat a
stream as invalid if it contains an ID header that does not have
enough data for these fields, even if it contain a valid 'magic
signature'. Future versions of this specification, even backwards-
compatible versions, might include additional fields in the ID
header. If an ID header has a compatible major version, but a larger
minor version, an implementation MUST NOT treat it as invalid for
containing additional data not specified here, provided it still
completes on the first page.
5.1.1. Channel Mapping
An Ogg Opus stream allows mapping one number of Opus streams (N) to a
possibly larger number of decoded channels (M + N) to yet another
number of output channels (C), which might be larger or smaller than
the number of decoded channels. The order and meaning of these
channels are defined by a channel mapping, which consists of the
'channel mapping family' octet and, for channel mapping families
other than family 0, a 'channel mapping table', as illustrated
in Figure 3.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+
| Stream Count |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Coupled Count | Channel Mapping... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: Channel Mapping Table
The fields in the channel mapping table have the following meaning:
1. Stream Count 'N' (8 bits, unsigned):
This is the total number of streams encoded in each Ogg packet.
This value is necessary to correctly parse the packed Opus
packets inside an Ogg packet, as described in Section 3. This
value MUST NOT be zero, as without at least one Opus packet with
a valid TOC sequence, a demuxer cannot recover the duration of an
Ogg packet.
For channel mapping family 0, this value defaults to 1, and is
not coded.
2. Coupled Stream Count 'M' (8 bits, unsigned):
This is the number of streams whose decoders are to be configured
to produce two channels (stereo). This MUST be no larger than
the total number of streams, N.
Each packet in an Opus stream has an internal channel count of 1
or 2, which can change from packet to packet. This is selected
by the encoder depending on the bitrate and the audio being
encoded. The original channel count of the audio passed to the
encoder is not necessarily preserved by the lossy compression.
Regardless of the internal channel count, any Opus stream can be
decoded as mono (a single channel) or stereo (two channels) by
appropriate initialization of the decoder. The 'coupled stream
count' field indicates that the decoders for the first M Opus
streams are to be initialized for stereo (two-channel) output,
and the remaining (N - M) decoders are to be initialized for mono
(a single channel) only. The total number of decoded channels,
(M + N), MUST be no larger than 255, as there is no way to index
more channels than that in the channel mapping.
For channel mapping family 0, this value defaults to (C - 1)
(i.e., 0 for mono and 1 for stereo), and is not coded.
3. Channel Mapping (8*C bits):
This contains one octet per output channel, indicating which
decoded channel is to be used for each one. Let 'index' be the
value of this octet for a particular output channel. This value
MUST either be smaller than (M + N) or be the special value 255.
If 'index' is less than 2*M, the output MUST be taken from
decoding stream ('index'/2) as stereo and selecting the left
channel if 'index' is even, and the right channel if 'index' is
odd. If 'index' is 2*M or larger, but less than 255, the output
MUST be taken from decoding stream ('index' - M) as mono. If
'index' is 255, the corresponding output channel MUST contain
pure silence.
The number of output channels, C, is not constrained to match the
number of decoded channels (M + N). A single index value MAY
appear multiple times, i.e., the same decoded channel might be
mapped to multiple output channels. Some decoded channels might
not be assigned to any output channel, as well.
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. There are three defined
mapping families in this specification.
5.1.1.1. Channel Mapping Family 0
Allowed numbers of channels: 1 or 2. RTP mapping. This is the same
channel interpretation as [RFC7587].
o 1 channel: monophonic (mono).
o 2 channels: stereo (left, right).
Special mapping: This channel mapping family also indicates that the
content consists of a single Opus stream that is stereo if and only
if C == 2, with stream index 0 mapped to output channel 0 (mono, or
left channel) and stream index 1 mapped to output channel 1 (right
channel) if stereo. When the 'channel mapping family' octet has this
value, the channel mapping table MUST be omitted from the ID header
packet.
5.1.1.2. Channel Mapping Family 1
Allowed numbers of channels: 1...8. Vorbis channel order (see
below).
Each channel is assigned to a speaker location in a conventional
surround arrangement. Specific locations depend on the number of
channels, and are given below in order of the corresponding channel
indices.
o 1 channel: monophonic (mono).
o 2 channels: stereo (left, right).
o 3 channels: linear surround (left, center, right).
o 4 channels: quadraphonic (front left, front right, rear left,
rear right).
o 5 channels: 5.0 surround (front left, front center, front right,
rear left, rear right).
o 6 channels: 5.1 surround (front left, front center, front right,
rear left, rear right, LFE).
o 7 channels: 6.1 surround (front left, front center, front right,
side left, side right, rear center, LFE).
o 8 channels: 7.1 surround (front left, front center, front right,
side left, side right, rear left, rear right, LFE).
This set of surround options and speaker location orderings is the
same as those used by the Vorbis codec [VORBIS-MAPPING]. The
ordering is different from the one used by the WAVE
[WAVE-MULTICHANNEL] and Free Lossless Audio Codec (FLAC) [FLAC]
formats, so correct ordering requires permutation of the output
channels when decoding to or encoding from those formats. "LFE" here
refers to a Low Frequency Effects channel, often mapped to a
subwoofer with no particular spatial position. Implementations
SHOULD identify "side" or "rear" speaker locations with "surround"
and "back" as appropriate when interfacing with audio formats or
systems that prefer that terminology.
5.1.1.3. Channel Mapping Family 255
Allowed numbers of channels: 1...255. No defined channel meaning.
Channels are unidentified. General-purpose players SHOULD NOT
attempt to play these streams. Offline implementations MAY
deinterleave the output into separate PCM files, one per channel.
Implementations SHOULD NOT produce output for channels mapped to
stream index 255 (pure silence) unless they have no other way to
indicate the index of non-silent channels.
5.1.1.4. Undefined Channel Mappings
The remaining channel mapping families (2...254) are reserved. A
demuxer implementation encountering a reserved 'channel mapping
family' value SHOULD act as though the value is 255.
5.1.1.5. Downmixing
An Ogg Opus player MUST support any valid channel mapping with a
channel mapping family of 0 or 1, even if the number of channels does
not match the physically connected audio hardware. Players SHOULD
perform channel mixing to increase or reduce the number of channels
as needed.
Implementations MAY use the matrices in Figures 4 through 9 to
implement downmixing from multichannel files using channel mapping
family 1 (Section 5.1.1.2), which are known to give acceptable
results for stereo. Matrices for 3 and 4 channels are normalized so
each coefficient row sums to 1 to avoid clipping. For 5 or more
channels, they are normalized to 2 as a compromise between clipping
and dynamic range reduction.
In these matrices the front-left and front-right channels are
generally passed through directly. When a surround channel is split
between both the left and right stereo channels, coefficients are
chosen so their squares sum to 1, which helps preserve the perceived
intensity. Rear channels are mixed more diffusely or attenuated to
maintain focus on the front channels.
L output = ( 0.585786 * left + 0.414214 * center )
R output = ( 0.414214 * center + 0.585786 * right )
Exact coefficient values are 1 and 1/sqrt(2), multiplied by
1/(1 + 1/sqrt(2)) for normalization.
Figure 4: Stereo Downmix Matrix for the
Linear Surround Channel Mapping
/ \ / \ / FL \
| L output | | 0.422650 0.000000 0.366025 0.211325 | | FR |
| R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL |
\ / \ / \ RR /
Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
1/(1 + sqrt(3)/2 + 1/2) for normalization.
Figure 5: Stereo Downmix Matrix for the Quadraphonic Channel Mapping
/ FL \
/ \ / \ | FC |
| L | | 0.650802 0.460186 0.000000 0.563611 0.325401 | | FR |
| R | = | 0.000000 0.460186 0.650802 0.325401 0.563611 | | RL |
\ / \ / | RR |
\ /
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2,
multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2) for normalization.
Figure 6: Stereo Downmix Matrix for the 5.0 Surround Mapping
/FL \
/ \ / \ |FC |
|L| | 0.529067 0.374107 0.000000 0.458186 0.264534 0.374107 | |FR |
|R| = | 0.000000 0.374107 0.529067 0.264534 0.458186 0.374107 | |RL |
\ / \ / |RR |
\LFE/
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2,
multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2)) for
normalization.
Figure 7: Stereo Downmix Matrix for the 5.1 Surround Mapping
/ \
| 0.455310 0.321953 0.000000 0.394310 0.227655 0.278819 0.321953 |
| 0.000000 0.321953 0.455310 0.227655 0.394310 0.278819 0.321953 |
\ /
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and
sqrt(3)/2/sqrt(2), multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 +
sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization. The coefficients
are in the same order as in Section 5.1.1.2 and the matrices above.
Figure 8: Stereo Downmix Matrix for the 6.1 Surround Mapping
/ \
| .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 |
| .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 |
\ /
Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2,
multiplied by 2/(2 + 2/sqrt(2) + sqrt(3)) for normalization. The
coefficients are in the same order as in Section 5.1.1.2 and the
matrices above.
Figure 9: Stereo Downmix Matrix for the 7.1 Surround Mapping
5.2. Comment Header
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'O' | 'p' | 'u' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 'T' | 'a' | 'g' | 's' |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
: Vendor String... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment List Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment #0 String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| |
: User Comment #0 String... :
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| User Comment #1 String Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: :
Figure 10: Comment Header Packet
The comment header consists of a 64-bit 'magic signature' field,
followed by data in the same format as the [VORBIS-COMMENT] header
used in Ogg Vorbis, except (like Ogg Theora and Speex) the final
'framing bit' specified in the Vorbis specification is not present.
1. Magic Signature:
This is an 8-octet (64-bit) field that allows codec
identification and is human readable. It contains, in order, the
magic numbers:
0x4F 'O'
0x70 'p'
0x75 'u'
0x73 's'
0x54 'T'
0x61 'a'
0x67 'g'
0x73 's'
Starting with "Op" helps distinguish it from audio data packets,
as this is an invalid TOC sequence.
2. Vendor String Length (32 bits, unsigned, little endian):
This field gives the length of the following vendor string, in
octets. It MUST NOT indicate that the vendor string is longer
than the rest of the packet.
3. Vendor String (variable length, UTF-8 vector):
This is a simple human-readable tag for vendor information,
encoded as a UTF-8 string [RFC3629]. No terminating null octet
is necessary.
This tag is intended to identify the codec encoder and
encapsulation implementations, for tracing differences in
technical behavior. User-facing applications can use the
'ENCODER' user comment tag to identify themselves.
4. User Comment List Length (32 bits, unsigned, little endian):
This field indicates the number of user-supplied comments. It
MAY indicate there are zero user-supplied comments, in which case
there are no additional fields in the packet. It MUST NOT
indicate that there are so many comments that the comment string
lengths would require more data than is available in the rest of
the packet.
5. User Comment #i String Length (32 bits, unsigned, little endian):
This field gives the length of the following user comment string,
in octets. There is one for each user comment indicated by the
'user comment list length' field. It MUST NOT indicate that the
string is longer than the rest of the packet.
6. User Comment #i String (variable length, UTF-8 vector):
This field contains a single user comment encoded as a UTF-8
string [RFC3629]. There is one for each user comment indicated
by the 'user comment list length' field.
The 'vendor string length' and 'user comment list length' fields are
REQUIRED, and implementations SHOULD treat a stream as invalid if it
contains a comment header that does not have enough data for these
fields, or that does not contain enough data for the corresponding
vendor string or user comments they describe. Making this check
before allocating the associated memory to contain the data helps
prevent a possible Denial-of-Service (DoS) attack from small comment
headers that claim to contain strings longer than the entire packet
or more user comments than could possibly fit in the packet.
Immediately following the user comment list, the comment header MAY
contain zero-padding or other binary data that is not specified here.
If the least-significant bit of the first byte of this data is 1,
then editors SHOULD preserve the contents of this data when updating
the tags, but if this bit is 0, all such data MAY be treated as
padding, and truncated or discarded as desired. This allows informal
experimentation with the format of this binary data until it can be
specified later.
The comment header can be arbitrarily large and might be spread over
a large number of Ogg pages. Implementations MUST avoid attempting
to allocate excessive amounts of memory when presented with a very
large comment header. To accomplish this, implementations MAY treat
a stream as invalid if it has a comment header larger than
125,829,120 octets (120 MB), and MAY ignore individual comments that
are not fully contained within the first 61,440 octets of the comment
header.
5.2.1. Tag Definitions
The user comment strings follow the NAME=value format described by
[VORBIS-COMMENT] with the same recommended tag names: ARTIST, TITLE,
DATE, ALBUM, and so on.
Two new comment tags are introduced here:
First, an optional gain for track normalization:
R128_TRACK_GAIN=-573
representing the volume shift needed to normalize the track's volume
during isolated playback, in random shuffle, and so on. The gain is
a Q7.8 fixed-point number in dB, as in the ID header's 'output gain'
field. This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in
Vorbis [REPLAY-GAIN], except that the normal volume reference is the
[EBU-R128] standard.
Second, an optional gain for album normalization:
R128_ALBUM_GAIN=111
representing the volume shift needed to normalize the overall volume
when played as part of a particular collection of tracks. The gain
is also a Q7.8 fixed-point number in dB, as in the ID header's
'output gain' field. The values '-573' and '111' given here are just
examples.
An Ogg Opus stream MUST NOT have more than one of each of these tags,
and, if present, their values MUST be an integer from -32768 to
32767, inclusive, represented in ASCII as a base 10 number with no
whitespace. A leading '+' or '-' character is valid. Leading zeros
are also permitted, but the value MUST be represented by no more than
6 characters. Other non-digit characters MUST NOT be present.
If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly
represent the R128 normalization gain relative to the 'output gain'
field specified in the ID header. If a player chooses to make use of
the R128_TRACK_GAIN tag or the R128_ALBUM_GAIN tag, it MUST apply
those gains _in addition_ to the 'output gain' value. If a tool
modifies the ID header's 'output gain' field, it MUST also update or
remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if
present. A muxer SHOULD place the gain it wants other tools to use
by default into the 'output gain' field, and not the comment tag.
To avoid confusion with multiple normalization schemes, an Opus
comment header SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN,
REPLAYGAIN_TRACK_PEAK, REPLAYGAIN_ALBUM_GAIN, or
REPLAYGAIN_ALBUM_PEAK tags, unless they are only to be used in some
context where there is guaranteed to be no such confusion.
[EBU-R128] normalization is preferred to the earlier REPLAYGAIN
schemes because of its clear definition and adoption by industry.
Peak normalizations are difficult to calculate reliably for lossy
codecs because of variation in excursion heights due to decoder
differences. In the authors' investigations, they were not applied
consistently or broadly enough to merit inclusion here.
6. Packet Size Limits
Technically, valid Opus packets can be arbitrarily large due to the
padding format, although the amount of non-padding data they can
contain is bounded. These packets might be spread over a similarly
enormous number of Ogg pages. When encoding, implementations SHOULD
limit the use of padding in audio data packets to no more than is
necessary to make a VBR stream CBR, unless they have no reasonable
way to determine what is necessary. Demuxers SHOULD treat audio data
packets as invalid (treat them as if they were malformed Opus packets
with an invalid TOC sequence) if they are larger than 61,440 octets
per Opus stream, unless they have a specific reason for allowing
extra padding. Such packets necessarily contain more padding than
needed to make a stream CBR. Demuxers MUST avoid attempting to
allocate excessive amounts of memory when presented with a very large
packet. Demuxers MAY treat audio data packets as invalid or
partially process them if they are larger than 61,440 octets in an
Ogg Opus stream with channel mapping families 0 or 1. Demuxers MAY
treat audio data packets as invalid or partially process them in any
Ogg Opus stream if the packet is larger than 61,440 octets and also
larger than 7,680 octets per Opus stream. The presence of an
extremely large packet in the stream could indicate a memory
exhaustion attack or stream corruption.
In an Ogg Opus stream, the largest possible valid packet that does
not use padding has a size of (61,298*N - 2) octets. With
255 streams, this is 15,630,988 octets and can span up to 61,298 Ogg
pages, all but one of which will have a granule position of -1. This
is, of course, a very extreme packet, consisting of 255 streams, each
containing 120 ms of audio encoded as 2.5 ms frames, each frame using
the maximum possible number of octets (1275) and stored in the least
efficient manner allowed (a VBR code 3 Opus packet). Even in such a
packet, most of the data will be zeros as 2.5 ms frames cannot
actually use all 1275 octets.
The largest packet consisting of entirely useful data is
(15,326*N - 2) octets. This corresponds to 120 ms of audio encoded
as 10 ms frames in either SILK or Hybrid mode, but at a data rate of
over 1 Mbps, which makes little sense for the quality achieved.
A more reasonable limit is (7,664*N - 2) octets. This corresponds to
120 ms of audio encoded as 20 ms stereo CELT mode frames, with a
total bitrate just under 511 kbps (not counting the Ogg encapsulation
overhead). For channel mapping family 1, N = 8 provides a reasonable
upper bound, as it allows for each of the 8 possible output channels
to be decoded from a separate stereo Opus stream. This gives a size
of 61,310 octets, which is rounded up to a multiple of 1,024 octets
to yield the audio data packet size of 61,440 octets that any
implementation is expected to be able to process successfully.
7. Encoder Guidelines
When encoding Opus streams, Ogg muxers SHOULD take into account the
algorithmic delay of the Opus encoder.
In encoders derived from the reference implementation [RFC6716], the
number of samples can be queried with
opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples));
To achieve good quality in the very first samples of a stream,
implementations MAY use linear predictive coding (LPC) extrapolation
to generate at least 120 extra samples at the beginning to avoid the
Opus encoder having to encode a discontinuous signal. For more
information on linear prediction, see [LINEAR-PREDICTION]. For an
input file containing 'length' samples, the implementation SHOULD set
the 'pre-skip' header value to (delay_samples + extra_samples),
encode at least (length + delay_samples + extra_samples) samples, and
set the granule position of the last page to
(length + delay_samples + extra_samples). This ensures that the
encoded file has the same duration as the original, with no time
offset. The best way to pad the end of the stream is to also use LPC
extrapolation, but zero-padding is also acceptable.
7.1. LPC Extrapolation
The first step in LPC extrapolation is to compute linear prediction
coefficients [LPC-SAMPLE]. When extending the end of the signal,
order-N (typically with N ranging from 8 to 40) LPC analysis is
performed on a window near the end of the signal. The last N samples
are used as memory to an infinite impulse response (IIR) filter.
The filter is then applied on a zero input to extrapolate the end of
the signal. Let 'a(k)' be the kth LPC coefficient and 'x(n)' be the
nth sample of the signal. Each new sample past the end of the signal
is computed as
N
---
x(n) = \ a(k)*x(n - k)
/
---
k = 1
The process is repeated independently for each channel. It is
possible to extend the beginning of the signal by applying the same
process backward in time. When extending the beginning of the
signal, it is best to apply a "fade in" to the extrapolated signal,
e.g., by multiplying it by a half-Hanning window [HANNING].
7.2. Continuous Chaining
In some applications, such as Internet radio, it is desirable to cut
a long stream into smaller chains, e.g., so the comment header can be
updated. This can be done simply by separating the input streams
into segments and encoding each segment independently. The drawback
of this approach is that it creates a small discontinuity at the
boundary due to the lossy nature of Opus. A muxer MAY avoid this
discontinuity by using the following procedure:
1. Encode the last frame of the first segment as an independent
frame by turning off all forms of inter-frame prediction.
De-emphasis is allowed.
2. Set the granule position of the last page to a point near the end
of the last frame.
3. Begin the second segment with a copy of the last frame of the
first segment.
4. Set the 'pre-skip' value of the second stream in such a way as to
properly join the two streams.
5. Continue the encoding process normally from there, without any
reset to the encoder.
In encoders derived from the reference implementation, inter-frame
prediction can be turned off by calling
opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));
For best results, this implementation requires that prediction be
explicitly enabled again before resuming normal encoding, even after
a reset.
8. Security Considerations
Implementations of the Opus codec need to take appropriate security
considerations into account, as outlined in [RFC4732]. This is just
as much a problem for the container as it is for the codec itself.
Malicious payloads and/or input streams can be used to attack codec
implementations. Implementations MUST NOT overrun their allocated
memory nor consume excessive resources when decoding payloads or
processing input streams. Although problems in encoding applications
are typically rarer, this still applies to a muxer, as
vulnerabilities would allow an attacker to attack transcoding
gateways.
Header parsing code contains the most likely area for potential
overruns. It is important for implementations to ensure their
buffers contain enough data for all of the required fields before
attempting to read it (for example, for all of the channel map data
in the ID header). Implementations would do well to validate the
indices of the channel map, also, to ensure they meet all of the
restrictions outlined in Section 5.1.1, in order to avoid attempting
to read data from channels that do not exist.
To avoid excessive resource usage, we advise implementations to be
especially wary of streams that might cause them to process far more
data than was actually transmitted. For example, a relatively small
comment header may contain values for the string lengths or user
comment list length that imply that it is many gigabytes in size.
Even computing the size of the required buffer could overflow a
32-bit integer, and actually attempting to allocate such a buffer
before verifying it would be a reasonable size is a bad idea. After
reading the user comment list length, implementations might wish to
verify that the header contains at least the minimum amount of data
for that many comments (4 additional octets per comment, to indicate
each has a length of zero) before proceeding any further, again
taking care to avoid overflow in these calculations. If allocating
an array of pointers to point at these strings, the size of the
pointers may be larger than 4 octets, potentially requiring a
separate overflow check.
Another bug in this class we have observed more than once involves
the handling of invalid data at the end of a stream. Often,
implementations will seek to the end of a stream to locate the last
timestamp in order to compute its total duration. If they do not
find a valid capture pattern and Ogg page from the desired logical
stream, they will back up and try again. If care is not taken to
avoid re-scanning data that was already scanned, this search can
quickly devolve into something with a complexity that is quadratic in
the amount of invalid data.
In general, when seeking, implementations will wish to be cautious
about the effects of invalid granule position values and ensure all
algorithms will continue to make progress and eventually terminate,
even if these are missing or out of order.
Like most other container formats, Ogg Opus streams SHOULD NOT be
used with insecure ciphers or cipher modes that are vulnerable to
known-plaintext attacks. Elements such as the Ogg page capture
pattern and the 'magic signature' fields in the ID header and the
comment header all have easily predictable values, in addition to
various elements of the codec data itself.
9. Content Type
An "Ogg Opus file" consists of one or more sequentially multiplexed
segments, each containing exactly one Ogg Opus stream. The
RECOMMENDED mime-type for Ogg Opus files is "audio/ogg".
If more specificity is desired, one MAY indicate the presence of Opus
streams using the codecs parameter defined in [RFC6381] and
[RFC5334], e.g.,
audio/ogg; codecs=opus
for an Ogg Opus file.
The RECOMMENDED filename extension for Ogg Opus files is '.opus'.
When Opus is concurrently multiplexed with other streams in an Ogg
container, one SHOULD use one of the "audio/ogg", "video/ogg", or
"application/ogg" mime-types, as defined in [RFC5334]. Such streams
are not strictly "Ogg Opus files" as described above, since they
contain more than a single Opus stream per sequentially multiplexed
segment, e.g., video or multiple audio tracks. In such cases, the
'.opus' filename extension is NOT RECOMMENDED.
In either case, this document updates [RFC5334] to add "opus" as a
codecs parameter value with char[8]: 'OpusHead' as Codec Identifier.
10. IANA Considerations
Per this document, IANA has updated the "Media Types" registry by
adding .opus as a file extension for "audio/ogg" and adding itself as
a reference alongside [RFC5334] for "audio/ogg", "video/ogg", and
"application/ogg" Media Types.
This document defines a new registry "Opus Channel Mapping Families"
to indicate how the semantic meanings of the channels in a multi-
channel Opus stream are described. IANA has created a new namespace
of "Opus Channel Mapping Families". This registry is listed on the
IANA Matrix. Modifications to this registry follow the
"Specification Required" registration policy as defined in [RFC5226].
Each registry entry consists of a Channel Mapping Family Number,
which is specified in decimal in the range 0 to 255, inclusive, and a
Reference (or list of references). Each Reference must point to
sufficient documentation to describe what information is coded in the
Opus identification header for this channel mapping family, how a
demuxer determines the stream count ('N') and coupled stream count
('M') from this information, and how it determines the proper
interpretation of each of the decoded channels.
This document defines three initial assignments for this registry.
+-------+---------------------------+
| Value | Reference |
+-------+---------------------------+
| 0 | RFC 7845, Section 5.1.1.1 |
| | |
| 1 | RFC 7845, Section 5.1.1.2 |
| | |
| 255 | RFC 7845, Section 5.1.1.3 |
+-------+---------------------------+
The designated expert will determine if the Reference points to a
specification that meets the requirements for permanence and ready
availability laid out in [RFC5226] and whether it specifies the
information described above with sufficient clarity to allow
interoperable implementations.
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
RFC 3533, DOI 10.17487/RFC3533, May 2003,
<https://www.rfc-editor.org/info/rfc3533>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <https://www.rfc-editor.org/info/rfc3629>.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
DOI 10.17487/RFC5226, May 2008,
<https://www.rfc-editor.org/info/rfc5226>.
[RFC5334] Goncalves, I., Pfeiffer, S., and C. Montgomery, "Ogg Media
Types", RFC 5334, DOI 10.17487/RFC5334, September 2008,
<https://www.rfc-editor.org/info/rfc5334>.
[RFC6381] Gellens, R., Singer, D., and P. Frojdh, "The 'Codecs' and
'Profiles' Parameters for "Bucket" Media Types", RFC 6381,
DOI 10.17487/RFC6381, August 2011,
<https://www.rfc-editor.org/info/rfc6381>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <https://www.rfc-editor.org/info/rfc6716>.
[EBU-R128]
EBU Technical Committee, "Loudness Recommendation EBU
R128", August 2011, <https://tech.ebu.ch/loudness>.
[VORBIS-COMMENT]
Montgomery, C., "Ogg Vorbis I Format Specification:
Comment Field and Header Specification", July 2002,
<https://www.xiph.org/vorbis/doc/v-comment.html>.
11.2. Informative References
[RFC4732] Handley, M., Ed., Rescorla, E., Ed., and IAB, "Internet
Denial-of-Service Considerations", RFC 4732,
DOI 10.17487/RFC4732, December 2006,
<https://www.rfc-editor.org/info/rfc4732>.
[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for the Opus Speech and Audio Codec", RFC 7587,
DOI 10.17487/RFC7587, June 2015,
<https://www.rfc-editor.org/info/rfc7587>.
[FLAC] Coalson, J., "FLAC - Free Lossless Audio Codec Format
Description", January 2008,
<https://xiph.org/flac/format.html>.
[HANNING] Wikipedia, "Hann window", February 2016,
<https://en.wikipedia.org/w/index.php?title=Window_functio
n&oldid=703074467#Hann_.28Hanning.29_window>.
[LINEAR-PREDICTION]
Wikipedia, "Linear Predictive Coding", October 2015,
<https://en.wikipedia.org/w/
index.php?title=Linear_predictive_coding&oldid=687498962>.
[LPC-SAMPLE]
Degener, J. and C. Bormann, "Autocorrelation LPC coeff
generation algorithm (Vorbis source code)", November 1994,
<https://svn.xiph.org/trunk/vorbis/lib/lpc.c>.
[Q-NOTATION]
Wikipedia, "Q (number format)", December 2015,
<https://en.wikipedia.org/w/
index.php?title=Q_%28number_format%29&oldid=697252615>.
[REPLAY-GAIN]
Parker, C. and M. Leese, "VorbisComment: Replay Gain",
June 2009,
<https://wiki.xiph.org/VorbisComment#Replay_Gain>.
[SEEKING] Pfeiffer, S., Parker, C., and G. Maxwell, "Granulepos
Encoding and How Seeking Really Works", May 2012,
<https://wiki.xiph.org/Seeking>.
[VORBIS-MAPPING]
Montgomery, C., "The Vorbis I Specification, Section 4.3.9
Output Channel Order", January 2010,
<https://www.xiph.org/vorbis/doc/
Vorbis_I_spec.html#x1-810004.3.9>.
[VORBIS-TRIM]
Montgomery, C., "The Vorbis I Specification, Appendix A:
Embedding Vorbis into an Ogg stream", November 2008,
<https://xiph.org/vorbis/doc/
Vorbis_I_spec.html#x1-132000A.2>.
[WAVE-MULTICHANNEL]
Microsoft Corporation, "Multiple Channel Audio Data and
WAVE Files", March 2007,
<https://msdn.microsoft.com/en-us/windows/hardware/
gg463006.aspx>.
Acknowledgments
Thanks to Ben Campbell, Joel M. Halpern, Mark Harris, Greg Maxwell,
Christopher "Monty" Montgomery, Jean-Marc Valin, Stephan Wenger, and
Mo Zanaty for their valuable contributions to this document.
Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent
Penquerc'h for their feedback based on early implementations.
Authors' Addresses
Timothy B. Terriberry
Mozilla Corporation
331 E. Evelyn Ave.
Mountain View, CA 94041
United States
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: +1 778 785 1540
Email: giles@xiph.org