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
April 2016 Ogg Encapsulation for the Opus Audio Codec
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
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described in the Simplified BSD License.
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
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
3. Packet Organization
An Ogg Opus stream is organized as follows (see Figure 1 for an
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
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
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
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
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
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
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
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.
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
'PCM sample position'
'playback time' = ---------------------
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).