codec T. Terriberry
Internet-Draft Mozilla Corporation
Intended status: Standards Track R. Lee
Expires: August 11, 2014 Voicetronix
R. Giles
Mozilla Corporation
February 7, 2014

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. 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

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This Internet-Draft will expire on August 11, 2014.

Copyright Notice

Copyright (c) 2014 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 ( 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

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 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].

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].

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".

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 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. 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 coding mode (SILK, Hybrid, or CELT), audio bandwidth, channel count, duration (frame size), and number of frames per packet, are indicated in the TOC (table of contents) in the first byte of each Opus packet, as described in Section 3.1 of [RFC6716]. The combination of mode, audio bandwidth, and frame size is referred to as the configuration of an Opus packet.

The first audio data page SHOULD NOT have the 'continued packet' flag set (which would indicate the first audio data packet is continued from a previous page). Packets MUST be placed into Ogg pages in order until the end of stream. Audio packets MAY span page boundaries. A decoder MUST treat a zero-octet audio data packet as if it were an Opus packet with an illegal TOC sequence. The last page SHOULD have the 'end of stream' flag set, but implementations 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'.

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.

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 muxer SHOULD emit packets that explicitly request the use of Packet Loss Concealment (PLC) in place of the missing packets. Only gaps that are a multiple of 2.5 ms are repairable, as these are the only durations that can be created by packet loss or discontinuous transmission. Muxers need not handle other gap sizes. Creating the necessary packets involves synthesizing a TOC byte (defined in Section 3.1 of [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 non-zero-length frames (using code 2 or 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 bytes 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 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 required 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 which 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 will also introduce latency (though the exact amount is not specified). Therefore, the first few samples produced by the decoder do not correspond to real input audio, but are instead composed of padding inserted by the encoder to compensate for this latency. These samples need to be stored and decoded, as Opus is an asymptotically convergent predictive codec, meaning the decoded contents of each frame depend on the recent history of decoder inputs. However, a decoder will want to skip these samples after decoding them.

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.

4.3. 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' = --------------------- .

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 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 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 MUST reject as invalid any stream where its 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. 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 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, and then continue to discard the samples required to reach the seek target (if any).

5. Header Packets

An Opus 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 1: ID Header Packet

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

To apply the gain, a decoder could use

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

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

  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, 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 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 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 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 next 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).

    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 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 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 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 audio 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 decoded as mono (a single channel) or stereo (two channels) by appropriate initialization of the decoder. The 'coupled stream count' field indicates that the first M Opus decoders are to be initialized for stereo output, and the remaining N-M decoders are to be initialized for mono only. The total number of decoded channels, (M+N), MUST be no larger than 255, as there is no way to index more channels than that in the channel mapping.

    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. Channel Mapping Family 0

Allowed numbers of channels: 1 or 2. RTP mapping.

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 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.

  • 1 channel: monophonic (mono).
  • 2 channels: stereo (left, right). Channel Mapping Family 1

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

Each channel is assigned to a speaker location in a conventional surround arrangement. Specific locations depend on the number of channels, and are given below in order of the corresponding channel indicies.

  • 1 channel: monophonic (mono).
  • 2 channels: stereo (left, right).
  • 3 channels: linear surround (left, center, right)
  • 4 channels: quadraphonic (front left, front right, rear left, rear right).
  • 5 channels: 5.0 surround (front left, front center, front right, rear left, rear right).
  • 6 channels: 5.1 surround (front left, front center, front right, rear left, rear right, LFE).
  • 7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE).
  • 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 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, often mapped to a subwoofer with no particular spatial position. Implementations SHOULD identify 'side' or 'rear' speaker locations with 'surround' and 'back' as appropriate when interfacing with audio formats or systems which prefer that terminology. 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, 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. Undefined Channel Mappings

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. Downmixing

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.

Implementations MAY use the following matricies to implement downmixing from multichannel files using Channel Mapping Family 1 [channel_mapping_1], which are known to give acceptable results for stereo. Matricies for 3 and 4 channels are normalized so each coefficent row sums to 1 to avoid clipping. For 5 or more channels they are normalized to 2 as a compromise between clipping and dynamic range reduction.

In these matricies the front left and front right channels are generally passed through directly. When a surround channel is split between both the left and right stereo channels, coefficients are chosen so their squares sum to 1, which helps preserve the perceived intensity. Rear channels are mixed more diffusely or attenuated to maintain focus on the front channels.

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 3: 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 4: 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 5: 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 |

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 6: 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 coeffients are in the same order as in Section, and the matricies above.

Figure 7: 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 coeffients are in the same order as in Section, and the matricies above.

Figure 8: 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 9: 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, except (like Ogg Theora and Speex) the final "framing bit" specified in the Vorbis spec 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 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.

  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 helps prevent a possible Denial-of-Service (DoS) attack from small comment headers that claim to contain strings longer than the entire packet or more user comments than than could possibly fit in the packet.

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:


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.

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.

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 cannot actually use all 1275 octets. The largest packet consisting of entirely useful data is (15,326*N - 2) octets, or about 15 kB per stream. This corresponds to 120 ms of audio encoded as 10 ms frames in either SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little sense for the quality achieved. A more reasonable limit is (7,664*N - 2) octets, or about 7.5 kB per stream. This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode frames, with a total bitrate just under 511 kbps (not counting the Ogg encapsulation overhead). With N=8, the maximum number of channels currently defined by mapping family 1, this gives a maximum packet size of 61,310 octets, or just under 60 kB. This is still quite conservative, as it assumes each output channel is taken from one decoded channel of a stereo packet. An implementation could reasonably choose any of these numbers for its internal limits.

7. Encoder Guidelines

When encoding Opus files, Ogg encoders should take into account the algorithmic delay of the Opus encoder.

In encoders derived from the reference implementation, 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, the Ogg encoder MAY use linear predictive coding (LPC) extrapolation [linear-prediction] to generate at least 120 extra samples at the beginning to avoid the Opus encoder having to encode a discontinuous signal. For an input file containing 'length' samples, the Ogg encoder SHOULD set the pre-skip header value to delay_samples+extra_samples, encode at least length+delay_samples+extra_samples samples, and set the granulepos of the last page to length+delay_samples+extra_samples. This ensures that the encoded file has the same duration as the original, with no time offset. The best way to pad the end of the stream is to also use LPC extrapolation, but zero-padding is also acceptable.

7.1. LPC Extrapolation

The first step in LPC extrapolation is to compute linear prediction coefficients. [lpc-sample] When extending the end of the signal, order-N (typically with N ranging from 8 to 40) LPC analysis is performed on a window near the end of the signal. The last N samples are used as memory to an infinite impulse response (IIR) filter.

The filter is then applied on a zero input to extrapolate the end of the signal. Let a(k) be the kth LPC coefficient and x(n) be the nth sample of the signal, each new sample past the end of the signal is computed as:

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

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. An encoder 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 granulepos 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);

Prediction should be enabled again before resuming normal encoding, even after a reset.

8. Implementation Status

A brief summary of major implementations of this draft is available at, along with their status.

[Note to RFC Editor: please remove this entire section before final publication per [RFC6982].]

9. 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.

10. 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], 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 the '.opus' filename extension is NOT RECOMMENDED.

11. IANA Considerations

This document has no actions for IANA.

12. Acknowledgments

Thanks to Greg Maxwell, Christopher "Monty" Montgomery, and Jean-Marc Valin for their valuable contributions to this document. Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penqeurc'h for their feedback based on early implementations.

13. 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.

14. References

14.1. Normative References

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0", RFC 3533, May 2003.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, November 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.
[RFC6716] Valin, JM., Vos, K. and T. Terriberry, "Definition of the Opus Audio Codec", RFC 6716, September 2012.
[EBU-R128] EBU Technical Committee, "Loudness Recommendation EBU R128", August 2011.
[vorbis-comment] Montgomery, C., "Ogg Vorbis I Format Specification: Comment Field and Header Specification", July 2002.

14.2. Informative References

[RFC4732] Handley, M., Rescorla, E. and IAB, "Internet Denial-of-Service Considerations", RFC 4732, December 2006.
[RFC6982] Sheffer, Y. and A. Farrel, "Improving Awareness of Running Code: The Implementation Status Section", RFC 6982, July 2013.
[flac] Coalson, J., "FLAC - Free Lossless Audio Codec Format Description", January 2008.
[hanning] Wikipedia, "Hann window", May 2013.
[linear-prediction] Wikipedia, "Linear Predictive Coding", January 2014.
[lpc-sample] Degener, J. and C. Bormann, "Autocorrelation LPC coeff generation algorithm (Vorbis source code)", November 1994.
[replay-gain] Parker, C. and M. Leese, "VorbisComment: Replay Gain", June 2009.
[seeking] Pfeiffer, S., Parker, C. and G. Maxwell, "Granulepos Encoding and How Seeking Really Works", May 2012.
[vorbis-mapping] Montgomery, C., "The Vorbis I Specification, Section 4.3.9 Output Channel Order", January 2010.
[vorbis-trim] Montgomery, C., "The Vorbis I Specification, Appendix A: Embedding Vorbis into an Ogg stream", November 2008.
[wave-multichannel] Microsoft Corporation, "Multiple Channel Audio Data and WAVE Files", March 2007.

Authors' Addresses

Timothy B. Terriberry Mozilla Corporation 650 Castro Street Mountain View, CA 94041 USA Phone: +1 650 903-0800 EMail:
Ron Lee Voicetronix 246 Pulteney Street, Level 1 Adelaide, SA 5000 Australia Phone: +61 8 8232 9112 EMail:
Ralph Giles Mozilla Corporation 163 West Hastings Street Vancouver, BC V6B 1H5 Canada Phone: +1 778 785 1540 EMail: