ref: a5bd4409319614f166a83934baddceed7e6c58b4
dir: /doc/draft-ietf-codec-oggopus.xml/
<?xml version="1.0" encoding="utf-8"?> <!DOCTYPE rfc SYSTEM 'rfc2629.dtd' [ <!ENTITY rfc2119 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.2119.xml'> <!ENTITY rfc3533 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.3533.xml'> <!ENTITY rfc3534 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.3534.xml'> <!ENTITY rfc3629 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.3629.xml'> <!ENTITY rfc4732 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.4732.xml'> <!ENTITY rfc6381 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.6381.xml'> <!ENTITY rfc6716 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.6716.xml'> ]> <?rfc toc="yes" symrefs="yes" ?> <rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-00"> <front> <title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title> <author initials="T.B." surname="Terriberry" fullname="Timothy B. Terriberry"> <organization>Mozilla Corporation</organization> <address> <postal> <street>650 Castro Street</street> <city>Mountain View</city> <region>CA</region> <code>94041</code> <country>USA</country> </postal> <phone>+1 650 903-0800</phone> <email>[email protected]</email> </address> </author> <author initials="R." surname="Lee" fullname="Ron Lee"> <organization>Voicetronix</organization> <address> <postal> <street>246 Pulteney Street, Level 1</street> <city>Adelaide</city> <region>SA</region> <code>5000</code> <country>Australia</country> </postal> <phone>+61 8 8232 9112</phone> <email>[email protected]</email> </address> </author> <author initials="R." surname="Giles" fullname="Ralph Giles"> <organization>Mozilla Corporation</organization> <address> <postal> <street>163 West Hastings Street</street> <city>Vancouver</city> <region>BC</region> <code>V6B 1H5</code> <country>Canada</country> </postal> <phone>+1 604 778 1540</phone> <email>[email protected]</email> </address> </author> <date day="19" month="November" year="2012"/> <area>RAI</area> <workgroup>codec</workgroup> <abstract> <t> This document defines the Ogg encapsulation for the Opus interactive speech and audio codec. This allows data encoded in the Opus format to be stored in an Ogg logical bitstream. Ogg encapsulation provides Opus with a long-term storage format supporting all of the essential features, including metadata, fast and accurate seeking, corruption detection, recapture after errors, low overhead, and the ability to multiplex Opus with other codecs (including video) with minimal buffering. It also provides a live streamable format, capable of delivery over a reliable stream-oriented transport, without requiring all the data, or even the total length of the data, up-front, in a form that is identical to the on-disk storage format. </t> </abstract> </front> <middle> <section anchor="intro" title="Introduction"> <t> The IETF Opus codec is a low-latency audio codec optimized for both voice and general-purpose audio. See <xref target="RFC6716"/> for technical details. This document defines the encapsulation of Opus in a continuous, logical Ogg bitstream <xref target="RFC3533"/>. </t> <t> Ogg bitstreams are made up of a series of 'pages', each of which contains data from one or more 'packets'. Pages are the fundamental unit of multiplexing in an Ogg stream. Each page is associated with a particular logical stream and contains a capture pattern and checksum, flags to mark the beginning and end of the logical stream, and a 'granule position' that represents an absolute position in the stream, to aid seeking. A single page can contain up to 65,025 octets of packet data from up to 255 different packets. Packets may be split arbitrarily across pages, and continued from one page to the next (allowing packets much larger than would fit on a single page). Each page contains 'lacing values' that indicate how the data is partitioned into packets, allowing a demuxer to recover the packet boundaries without examining the encoded data. A packet is said to 'complete' on a page when the page contains the final lacing value corresponding to that packet. </t> <t> This encapsulation defines the 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 <xref target="RFC3533"/>. </t> </section> <section anchor="terminology" title="Terminology"> <t> The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in <xref target="RFC2119"/>. </t> <t> Implementations that fail to satisfy one or more "MUST" requirements are considered non-compliant. Implementations that satisfy all "MUST" requirements, but fail to satisfy one or more "SHOULD" requirements are said to be "conditionally compliant". All other implementations are "unconditionally compliant". </t> </section> <section anchor="packet_organization" title="Packet Organization"> <t> An Opus stream is organized as follows. </t> <t> There are two mandatory header packets. The granule position of the pages on which these packets complete MUST be zero. </t> <t> The first packet in the logical Ogg bitstream MUST contain the identification (ID) header, which uniquely identifies a stream as Opus audio. The format of this header is defined in <xref target="id_header"/>. It MUST be placed alone (without any other packet data) on the first page of the logical Ogg bitstream, and must complete on that page. This page MUST have its 'beginning of stream' flag set. </t> <t> The second packet in the logical Ogg bitstream MUST contain the comment header, which contains user-supplied metadata. The format of this header is defined in <xref target="comment_header"/>. It MAY span one or more pages, beginning on the second page of the logical stream. However many pages it spans, the comment header packet MUST finish the page on which it completes. </t> <t> All subsequent pages are audio data pages, and the Ogg packets they contain are audio data packets. Each audio data packet contains one Opus packet for each of N different streams, where N is typically one for mono or stereo, but may be greater than one for, e.g., multichannel audio. The value N is specified in the ID header (see <xref target="channel_mapping"/>), and is fixed over the entire length of the logical Ogg bitstream. </t> <t> The first N-1 Opus packets, if any, are packed one after another into the Ogg packet, using the self-delimiting framing from Appendix B of <xref target="RFC6716"/>. The remaining Opus packet is packed at the end of the Ogg packet using the regular, undelimited framing from Section 3 of <xref target="RFC6716"/>. All of the Opus packets in a single Ogg packet MUST be constrained to have the same duration. The duration and coding modes of each Opus packet are contained in the TOC (table of contents) sequence in the first few bytes. A decoder SHOULD treat any Opus packet whose duration is different from that of the first Opus packet in an Ogg packet as if it were an Opus packet with an illegal TOC sequence. </t> <t> The first audio data page SHOULD NOT have the 'continued packet' flag set (which would indicate the first audio data packet is continued from a previous page). Packets MUST be placed into Ogg pages in order until the end of stream. Audio packets MAY span page boundaries. A decoder MUST treat a zero-octet audio data packet as if it were 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'. </t> </section> <section anchor="granpos" title="Granule Position"> <t> The granule position of an audio data page encodes the total number of PCM samples in the stream up to and including the last fully-decodable sample from the last packet completed on that page. A page that is entirely spanned by a single packet (that completes on a subsequent page) has no granule position, and the granule position field MUST be set to the special value '-1' in two's complement. </t> <t> The granule position of an audio data page is in units of PCM audio samples at a fixed rate of 48 kHz (per channel; a stereo stream's granule position does not increment at twice the speed of a mono stream). It is possible to run an Opus decoder at other sampling rates, but the value in the granule position field always counts samples assuming a 48 kHz decoding rate, and the rest of this specification makes the same assumption. </t> <t> The duration of an Opus packet 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. </t> <t> All other pages with completed packets after the first MUST have a granule position equal to the number of samples contained in packets that complete on that page plus the granule position of the most recent page with completed packets. This guarantees that a demuxer can assign individual packets the same granule position when working forwards as when working backwards. For this to work, there cannot be any gaps. In order to support capturing a stream that uses discontinuous transmission (DTX), an encoder SHOULD emit packets that explicitly request the use of Packet Loss Concealment (PLC) (i.e., with a frame length of 0, as defined in Section 3.2.1 of <xref target="RFC6716"/>) in place of the packets that were not transmitted. </t> <section anchor="preskip" title="Pre-skip"> <t> There is some amount of latency introduced during the decoding process, to allow for overlap in the MDCT modes, stereo mixing in the LP modes, and resampling, and the encoder will introduce even more latency (though the exact amount is not specified). Therefore, the first few samples produced by the decoder do not correspond to real input audio, but are instead composed of padding inserted by the encoder to compensate for this latency. These samples need to be stored and decoded, as Opus is an asymptotically convergent predictive codec, meaning the decoded contents of each frame depend on the recent history of decoder inputs. However, a decoder will want to skip these samples after decoding them. </t> <t> A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals the number of samples which SHOULD be skipped (decoded but discarded) at the beginning of the stream. This 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. </t> </section> <section anchor="pcm_sample_position" title="PCM Sample Position"> <t> The PCM sample position is determined from the granule position using the formula <figure align="center"> <artwork align="center"><![CDATA[ 'PCM sample position' = 'granule position' - 'pre-skip' . ]]></artwork> </figure> </t> <t> For example, if the granule position of the first audio data page is 59,971, and the pre-skip is 11,971, then the PCM sample position of the last decoded sample from that page is 48,000. This can be converted into a playback time using the formula <figure align="center"> <artwork align="center"><![CDATA[ 'PCM sample position' 'playback time' = --------------------- . 48000.0 ]]></artwork> </figure> </t> <t> The initial PCM sample position before any samples are played is normally '0'. In this case, the PCM sample position of the first audio sample to be played starts at '1', because it marks the time on the clock <spanx style="emph">after</spanx> that sample has been played, and a stream that is exactly one second long has a final PCM sample position of '48000', as in the example here. </t> <t> Vorbis streams use a granule position smaller than the number of audio samples contained in the first audio data page to indicate that some of those samples must be trimmed from the output (see <xref target="vorbis-trim"/>). However, to do so, Vorbis requires that the first audio data page contains exactly two packets, in order to allow the decoder to perform PCM position adjustments before needing to return any PCM data. Opus uses the pre-skip mechanism for this purpose instead, since the encoder may introduce more than a single packet's worth of latency, and since very large packets in streams with a very large number of channels might not fit on a single page. </t> </section> <section anchor="end_trimming" title="End Trimming"> <t> The page with the 'end of stream' flag set MAY have a granule position that indicates the page contains less audio data than would normally be returned by decoding up through the final packet. This is used to end the stream somewhere other than an even frame boundary. The granule position of the most recent audio data page with completed packets is used to make this determination, or '0' is used if there were no previous audio data pages with a completed packet. The difference between these granule positions indicates how many samples to keep after decoding the packets that completed on the final page. The remaining samples are discarded. The number of discarded samples SHOULD be no larger than the number decoded from the last packet. </t> </section> <section anchor="start_granpos_restrictions" title="Restrictions on the Initial Granule Position"> <t> The granule position of the first audio data page with a completed packet MAY be larger than the number of samples contained in packets that complete on that page, however it MUST NOT be smaller, unless that page has the 'end of stream' flag set. Allowing a granule position larger than the number of samples allows the beginning of a stream to be cropped or a live stream to be joined without rewriting the granule position of all the remaining pages. This means that the PCM sample position just before the first sample to be played may be larger than '0'. Synchronization when multiplexing with other logical streams still uses the PCM sample position relative to '0' to compute sample times. This does not affect the behavior of pre-skip: exactly 'pre-skip' samples should be skipped from the beginning of the decoded output, even if the initial PCM sample position is greater than zero. </t> <t> On the other hand, a granule position that is smaller than the number of decoded samples prevents a demuxer from working backwards to assign each packet or each individual sample a valid granule position, since granule positions 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. </t> <t> If that page has the 'end of stream' flag set, a demuxer MUST reject as invalid any stream where its granule position is smaller than the 'pre-skip' amount. This would indicate that 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. </t> </section> <section anchor="seeking_and_preroll" title="Seeking and Pre-roll"> <t> Seeking in Ogg files is best performed using a bisection search for a page whose granule position corresponds to a PCM position at or before the seek target. With appropriately weighted bisection, accurate seeking can be performed with just three or four bisections even in multi-gigabyte files. See <xref target="seeking"/> for general implementation guidance. </t> <t> When seeking within an Ogg Opus stream, the decoder SHOULD start decoding (and discarding the output) at least 3840 samples (80 ms) prior to the seek target in order to ensure that the output audio is correct by the time it reaches the seek target. This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the beginning of the stream. If the point 80 ms prior to the seek target comes before the initial PCM sample position, the decoder SHOULD start decoding from the beginning of the stream, applying pre-skip as normal, regardless of whether the pre-skip is larger or smaller than 80 ms, and then continue to discard the samples required to reach the seek target (if any). </t> </section> </section> <section anchor="headers" title="Header Packets"> <t> An Opus stream contains exactly two mandatory header packets. </t> <section anchor="id_header" title="Identification Header"> <figure anchor="id_header_packet" title="ID Header Packet" align="center"> <artwork align="center"><![CDATA[ 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'O' | 'p' | 'u' | 's' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'H' | 'e' | 'a' | 'd' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Version = 1 | Channel Count | Pre-skip | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Input Sample Rate (Hz) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Output Gain (Q7.8 in dB) | Mapping Family| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : | | : Optional Channel Mapping Table... : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ]]></artwork> </figure> <t> The fields in the identification (ID) header have the following meaning: <list style="numbers"> <t><spanx style="strong">Magic Signature</spanx>: <vspace blankLines="1"/> This is an 8-octet (64-bit) field that allows codec identification and is human-readable. It contains, in order, the magic numbers: <list style="empty"> <t>0x4F 'O'</t> <t>0x70 'p'</t> <t>0x75 'u'</t> <t>0x73 's'</t> <t>0x48 'H'</t> <t>0x65 'e'</t> <t>0x61 'a'</t> <t>0x64 'd'</t> </list> Starting with "Op" helps distinguish it from audio data packets, as this is an invalid TOC sequence. <vspace blankLines="1"/> </t> <t><spanx style="strong">Version</spanx> (8 bits, unsigned): <vspace blankLines="1"/> The version number MUST always be '1' for this version of the encapsulation specification. Implementations SHOULD treat streams where the upper four bits of the version number match that of a recognized specification as backwards-compatible with that specification. That is, the version number can be split into "major" and "minor" version sub-fields, with changes to the "minor" sub-field (in the lower four bits) signaling compatible changes. For example, a decoder implementing this specification SHOULD accept any stream with a version number of '15' or less, and SHOULD assume any stream with a version number '16' or greater is incompatible. The initial version '1' was chosen to keep implementations from relying on this octet as a null terminator for the "OpusHead" string. <vspace blankLines="1"/> </t> <t><spanx style="strong">Output Channel Count</spanx> 'C' (8 bits, unsigned): <vspace blankLines="1"/> This is the number of output channels. This might be different than the number of encoded channels, which can change on a packet-by-packet basis. This value MUST NOT be zero. The maximum allowable value depends on the channel mapping family, and might be as large as 255. See <xref target="channel_mapping"/> for details. <vspace blankLines="1"/> </t> <t><spanx style="strong">Pre-skip</spanx> (16 bits, unsigned, little endian): <vspace blankLines="1"/> This is the number of samples (at 48 kHz) to discard from the decoder output when starting playback, and also the number to subtract from a page's granule position to calculate its PCM sample position. When constructing cropped Ogg Opus streams, a pre-skip of at least 3,840 samples (80 ms) is RECOMMENDED to ensure complete convergence. <vspace blankLines="1"/> </t> <t><spanx style="strong">Input Sample Rate</spanx> (32 bits, unsigned, little endian): <vspace blankLines="1"/> This field is <spanx style="emph">not</spanx> the sample rate to use for playback of the encoded data. <vspace blankLines="1"/> Opus has a handful of coding modes, with internal audio bandwidths of 4, 6, 8, 12, and 20 kHz. Each packet in the stream may have a different audio bandwidth. Regardless of the audio bandwidth, the reference decoder supports decoding any stream at a sample rate of 8, 12, 16, 24, or 48 kHz. The original sample rate of the encoder input is not preserved by the lossy compression. <vspace blankLines="1"/> An Ogg Opus player SHOULD select the playback sample rate according to the following procedure: <list style="numbers"> <t>If the hardware supports 48 kHz playback, decode at 48 kHz.</t> <t>Otherwise, if the hardware's highest available sample rate is a supported rate, decode at this sample rate.</t> <t>Otherwise, if the hardware's highest available sample rate is less than 48 kHz, decode at the highest supported rate above this and resample.</t> <t>Otherwise, decode at 48 kHz and resample.</t> </list> However, the 'Input Sample Rate' field allows the encoder to pass the sample rate of the original input stream as metadata. This 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. <vspace blankLines="1"/> A value of zero indicates 'unspecified'. Encoders SHOULD write the actual input sample rate or zero, but decoder implementations which do something with this field SHOULD take care to behave sanely if given crazy values (e.g., do not actually upsample the output to 10 MHz if requested). <vspace blankLines="1"/> </t> <t><spanx style="strong">Output Gain</spanx> (16 bits, signed, little endian): <vspace blankLines="1"/> This is a gain to be applied by the decoder. It is 20*log10 of the factor to scale the decoder output by to achieve the desired playback volume, stored in a 16-bit, signed, two's complement fixed-point value with 8 fractional bits (i.e., Q7.8). To apply the gain, a decoder could use <figure align="center"> <artwork align="center"><![CDATA[ sample *= pow(10, output_gain/(20.0*256)) , ]]></artwork> </figure> where output_gain is the raw 16-bit value from the header. <vspace blankLines="1"/> Virtually all players and media frameworks should apply it by default. If a player chooses to apply any volume adjustment or gain modification, such as the R128_TRACK_GAIN (see <xref target="comment_header"/>) 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. <vspace blankLines="1"/> An encoder SHOULD set this field to zero, and instead apply any gain prior to encoding, when this is possible and does not conflict with the user's wishes. 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. <vspace blankLines="1"/> Although the output gain has enormous range (+/- 128 dB, enough to amplify inaudible sounds to the threshold of physical pain), most applications can only reasonably use a small portion of this range around zero. The large range serves in part to ensure that gain can always be losslessly transferred between OpusHead and R128_TRACK_GAIN (see below) without saturating. <vspace blankLines="1"/> </t> <t><spanx style="strong">Channel Mapping Family</spanx> (8 bits, unsigned): <vspace blankLines="1"/> This octet indicates the order and semantic meaning of the various channels encoded in each Ogg packet. <vspace blankLines="1"/> Each possible value of this octet indicates a mapping family, which defines a set of allowed channel counts, and the ordered set of channel names for each allowed channel count. The details are described in <xref target="channel_mapping"/>. </t> <t><spanx style="strong">Channel Mapping Table</spanx>: This table defines the mapping from encoded streams to output channels. It is omitted when the channel mapping family is 0, but REQUIRED otherwise. Its contents are specified in <xref target="channel_mapping"/>. </t> </list> </t> <t> All fields in the ID headers are REQUIRED, except for the channel mapping table, which is omitted when the channel mapping family is 0. Implementations SHOULD reject ID headers which do not contain enough data for these fields, even if they contain a valid Magic Signature. Future versions of this specification, even backwards-compatible versions, might include additional fields in the ID header. If an ID header has a compatible major version, but a larger minor version, an implementation MUST NOT reject it for containing additional data not specified here. However, implementations MAY reject streams in which the ID header does not complete on the first page. </t> <section anchor="channel_mapping" title="Channel Mapping"> <t> An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly larger number of decoded channels (M+N) to yet another number of output channels (C), which might be larger or smaller than the number of decoded channels. The order and meaning of these channels are defined by a channel mapping, which consists of the 'channel mapping family' octet and, for channel mapping families other than family 0, a channel mapping table, as illustrated in <xref target="channel_mapping_table"/>. </t> <figure anchor="channel_mapping_table" title="Channel Mapping Table" align="center"> <artwork align="center"><![CDATA[ 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+ | Stream Count | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Coupled Count | Channel Mapping... : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ]]></artwork> </figure> <t> The fields in the channel mapping table have the following meaning: <list style="numbers" counter="8"> <t><spanx style="strong">Stream Count</spanx> 'N' (8 bits, unsigned): <vspace blankLines="1"/> This is the total number of streams encoded in each Ogg packet. This value is required to correctly parse the packed Opus packets inside an Ogg packet, as described in <xref target="packet_organization"/>. This value MUST NOT be zero, as without at least one Opus packet with a valid TOC sequence, a demuxer cannot recover the duration of an Ogg packet. <vspace blankLines="1"/> For channel mapping family 0, this value defaults to 1, and is not coded. <vspace blankLines="1"/> </t> <t><spanx style="strong">Coupled Stream Count</spanx> 'M' (8 bits, unsigned): This is the number of streams whose decoders should be configured to produce two channels. This MUST be no larger than the total number of streams, N. <vspace blankLines="1"/> Each packet in an Opus stream has an internal channel count of 1 or 2, which can change from packet to packet. This is selected by the encoder depending on the bitrate and the contents being encoded. The original channel count of the encoder input is not preserved by the lossy compression. <vspace blankLines="1"/> Regardless of the internal channel count, any Opus stream can be decoded as mono (a single channel) or stereo (two channels) by appropriate initialization of the decoder. The 'coupled stream count' field indicates that the first M Opus decoders are to be initialized in stereo mode, and the remaining N-M decoders are to be initialized in mono mode. The total number of decoded channels, (M+N), MUST be no larger than 255, as there is no way to index more channels than that in the channel mapping. <vspace blankLines="1"/> For channel mapping family 0, this value defaults to C-1 (i.e., 0 for mono and 1 for stereo), and is not coded. <vspace blankLines="1"/> </t> <t><spanx style="strong">Channel Mapping</spanx> (8*C bits): This contains one octet per output channel, indicating which decoded channel 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. <vspace blankLines="1"/> The number of output channels, C, is not constrained to match the number of decoded channels (M+N). A single index value MAY appear multiple times, i.e., the same decoded channel might be mapped to multiple output channels. Some decoded channels might not be assigned to any output channel, as well. <vspace blankLines="1"/> For channel mapping family 0, the first index defaults to 0, and if C==2, the second index defaults to 1. Neither index is coded. </t> </list> </t> <t> After producing the output channels, the channel mapping family determines the semantic meaning of each one. Currently there are three defined mapping families, although more may be added: <list style="symbols"> <t>Family 0 (RTP mapping): <vspace blankLines="1"/> Allowed numbers of channels: 1 or 2. <list style="symbols"> <t>1 channel: monophonic (mono).</t> <t>2 channels: stereo (left, right).</t> </list> <spanx style="strong">Special mapping</spanx>: This channel mapping value also indicates that the contents consists of a single Opus stream that is stereo if and only if C==2, with stream index 0 mapped to channel 0, and (if stereo) stream index 1 mapped to channel 1. When the 'channel mapping family' octet has this value, the channel mapping table MUST be omitted from the ID header packet. <vspace blankLines="1"/> </t> <t>Family 1 (Vorbis channel order): <vspace blankLines="1"/> Allowed numbers of channels: 1...8.<vspace/> Channel meanings depend on the number of channels. See <xref target="vorbis-mapping"/> for the assignments from output channel number to specific speaker locations. <vspace blankLines="1"/> </t> <t>Family 255 (no defined channel meaning): <vspace blankLines="1"/> Allowed numbers of channels: 1...255.<vspace/> Channels are unidentified. General-purpose players SHOULD NOT attempt to play these streams, and offline decoders MAY deinterleave the output into separate PCM files, one per channel. Decoders SHOULD NOT produce output for channels mapped to stream index 255 (pure silence) unless they have no other way to indicate the index of non-silent channels. </t> </list> 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. <vspace blankLines="1"/> An Ogg Opus player MUST play any Ogg Opus stream with a channel mapping family of 0 or 1, even if the number of channels does not match the physically connected audio hardware. Players SHOULD perform channel mixing to increase or reduce the number of channels as needed. </t> </section> </section> <section anchor="comment_header" title="Comment Header"> <figure anchor="comment_header_packet" title="Comment Header Packet" align="center"> <artwork align="center"><![CDATA[ 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'O' | 'p' | 'u' | 's' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'T' | 'a' | 'g' | 's' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Vendor String Length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | : Vendor String... : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | User Comment List Length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | User Comment #0 String Length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | : User Comment #0 String... : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | User Comment #1 String Length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : : ]]></artwork> </figure> <t> The comment header consists of a 64-bit magic signature, followed by data in the same format as the <xref target="vorbis-comment"/> header used in Ogg Vorbis (without the final "framing bit"), Ogg Theora, and Speex. <list style="numbers"> <t><spanx style="strong">Magic Signature</spanx>: <vspace blankLines="1"/> This is an 8-octet (64-bit) field that allows codec identification and is human-readable. It contains, in order, the magic numbers: <list style="empty"> <t>0x4F 'O'</t> <t>0x70 'p'</t> <t>0x75 'u'</t> <t>0x73 's'</t> <t>0x54 'T'</t> <t>0x61 'a'</t> <t>0x67 'g'</t> <t>0x73 's'</t> </list> Starting with "Op" helps distinguish it from audio data packets, as this is an invalid TOC sequence. <vspace blankLines="1"/> </t> <t><spanx style="strong">Vendor String Length</spanx> (32 bits, unsigned, little endian): <vspace blankLines="1"/> This field gives the length of the following vendor string, in octets. It MUST NOT indicate that the vendor string is longer than the rest of the packet. <vspace blankLines="1"/> </t> <t><spanx style="strong">Vendor String</spanx> (variable length, UTF-8 vector): <vspace blankLines="1"/> This is a simple human-readable tag for vendor information, encoded as a UTF-8 string <xref target="RFC3629"/>. No terminating NUL octet is required. <vspace blankLines="1"/> This tag is intended to identify the codec encoder and encapsulation implementations, for tracing differences in technical behavior. User-facing encoding applications can use the 'ENCODER' user comment tag to identify themselves. <vspace blankLines="1"/> </t> <t><spanx style="strong">User Comment List Length</spanx> (32 bits, unsigned, little endian): <vspace blankLines="1"/> This field indicates the number of user-supplied comments. It MAY indicate there are zero user-supplied comments, in which case there are no additional fields in the packet. It MUST NOT indicate that there are so many comments that the comment string lengths would require more data than is available in the rest of the packet. <vspace blankLines="1"/> </t> <t><spanx style="strong">User Comment #i String Length</spanx> (32 bits, unsigned, little endian): <vspace blankLines="1"/> This field gives the length of the following user comment string, in octets. There is one for each user comment indicated by the 'user comment list length' field. It MUST NOT indicate that the string is longer than the rest of the packet. <vspace blankLines="1"/> </t> <t><spanx style="strong">User Comment #i String</spanx> (variable length, UTF-8 vector): <vspace blankLines="1"/> This field contains a single user comment string. There is one for each user comment indicated by the 'user comment list length' field. </t> </list> </t> <t> The vendor string length and user comment list length are REQUIRED, and implementations SHOULD reject comment headers that do not contain enough data for these fields, or that do not contain enough data for the corresponding vendor string or user comments they describe. Making this check before allocating the associated memory to contain the data may help prevent a possible Denial-of-Service (DoS) attack from small comment headers that claim to contain strings longer than the entire packet or more user comments than than could possibly fit in the packet. </t> <t> The user comment strings follow the NAME=value format described by <xref target="vorbis-comment"/> with the same recommended tag names. One new comment tag is introduced for Ogg Opus: <figure align="center"> <artwork align="left"><![CDATA[ R128_TRACK_GAIN=-573 ]]></artwork> </figure> 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 <xref target="replay-gain"/>, except that the normal volume reference is the <xref target="EBU-R128"/> standard. </t> <t> 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 <spanx style="emph">in addition</spanx> 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. </t> <t> To avoid confusion with multiple normalization schemes, an Opus comment header SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK, REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags. </t> <t> 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. </t> </section> </section> <section anchor="packet_size_limits" title="Packet Size Limits"> <t> Technically valid Opus packets can be arbitrarily large due to the padding format, although the amount of non-padding data they can contain is bounded. These packets might be spread over a similarly enormous number of Ogg pages. Encoders SHOULD use no more padding than 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. </t> <t> In an Ogg Opus stream, the largest possible valid packet that does not use padding has a size of (61,298*N - 2) octets, or about 60 kB per Opus stream. With 255 streams, this is 15,630,988 octets (14.9 MB) and can span up to 61,298 Ogg pages, all but one of which will have a granule position of -1. This is of course a very extreme packet, consisting of 255 streams, each containing 120 ms of audio encoded as 2.5 ms frames, each frame using the maximum possible number of octets (1275) and stored in the least efficient manner allowed (a VBR code 3 Opus packet). Even in such a packet, most of the data will be zeros, as 2.5 ms frames, which are required to run in the MDCT mode, cannot actually use all 1275 octets. The largest packet consisting of entirely useful data is (15,326*N - 2) octets, or about 15 kB per stream. This corresponds to 120 ms of audio encoded as 10 ms frames in either LP or Hybrid mode, but at a data rate of over 1 Mbps, which makes little sense for the quality achieved. A more reasonable limit is (7,664*N - 2) octets, or about 7.5 kB per stream. This corresponds to 120 ms of audio encoded as 20 ms stereo MDCT-mode frames, with a total bitrate just under 511 kbps (not counting the Ogg encapsulation overhead). With N=8, the maximum number of channels currently defined by mapping family 1, this gives a maximum packet size of 61,310 octets, or just under 60 kB. This is still quite conservative, as it assumes each output channel is taken from one decoded channel of a stereo packet. An implementation could reasonably choose any of these numbers for its internal limits. </t> </section> <section anchor="security" title="Security Considerations"> <t> Implementations of the Opus codec need to take appropriate security considerations into account, as outlined in <xref target="RFC4732"/>. This is just as much a problem for the container as it is for the codec itself. It is extremely important for the decoder to be robust against malicious payloads. Malicious payloads must not cause the decoder to overrun its allocated memory or to take an excessive amount of resources to decode. Although problems in encoders are typically rarer, the same applies to the encoder. Malicious audio streams must not cause the encoder to misbehave because this would allow an attacker to attack transcoding gateways. </t> <t> Like most other container formats, Ogg Opus 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. </t> </section> <section anchor="content_type" title="Content Type"> <t> An "Ogg Opus file" consists of one or more sequentially multiplexed segments, each containing exactly one Ogg Opus stream. The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg". When Opus is concurrently multiplexed with other streams in an Ogg container, one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg" mime-types, as defined in <xref target="RFC3534"/>. </t> <t> If more specificity is desired, one MAY indicate the presence of Opus streams using the codecs parameter defined in <xref target="RFC6381"/>, e.g., <figure align="center"> <artwork align="left"><![CDATA[ audio/ogg; codecs=opus ]]></artwork> </figure> for an Ogg Opus file. </t> <t> The RECOMMENDED filename extension for Ogg Opus files is '.opus'. </t> </section> <section title="IANA Considerations"> <t> This document has no actions for IANA. </t> </section> <section anchor="Acknowledgments" title="Acknowledgments"> <t> Thanks to Greg Maxwell, Christopher "Monty" Montgomery, and Jean-Marc Valin for their valuable contributions to this document. Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penqeurc'h for their feedback based on early implementations. </t> </section> <section title="Copying Conditions"> <t> The authors agree to grant third parties the irrevocable right to copy, use, and distribute the work, with or without modification, in any medium, without royalty, provided that, unless separate permission is granted, redistributed modified works do not contain misleading author, version, name of work, or endorsement information. </t> </section> </middle> <back> <references title="Normative References"> &rfc2119; &rfc3533; &rfc3534; &rfc3629; &rfc6381; &rfc6716; <reference anchor="EBU-R128" target="http://tech.ebu.ch/loudness"> <front> <title>"Loudness Recommendation EBU R128</title> <author fullname="EBU Technical Committee"/> <date month="August" year="2011"/> </front> </reference> <reference anchor="vorbis-comment" target="http://www.xiph.org/vorbis/doc/v-comment.html"> <front> <title>Ogg Vorbis I Format Specification: Comment Field and Header Specification</title> <author initials="C." surname="Montgomery" fullname="Christopher "Monty" Montgomery"/> <date month="July" year="2002"/> </front> </reference> <reference anchor="vorbis-mapping" target="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9"> <front> <title>The Vorbis I Specification, Section 4.3.9 Output Channel Order</title> <author initials="C." surname="Montgomery" fullname="Christopher "Monty" Montgomery"/> <date month="January" year="2010"/> </front> </reference> </references> <references title="Informative References"> <!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml"?--> &rfc4732; <reference anchor="replay-gain" target="http://wiki.xiph.org/VorbisComment#Replay_Gain"> <front> <title>VorbisComment: Replay Gain</title> <author initials="C." surname="Parker" fullname="Conrad Parker"/> <author initials="M." surname="Leese" fullname="Martin Leese"/> <date month="June" year="2009"/> </front> </reference> <reference anchor="seeking" target="http://wiki.xiph.org/Seeking"> <front> <title>Granulepos Encoding and How Seeking Really Works</title> <author initials="S." surname="Pfeiffer" fullname="Silvia Pfeiffer"/> <author initials="C." surname="Parker" fullname="Conrad Parker"/> <author initials="G." surname="Maxwell" fullname="Greg Maxwell"/> <date month="May" year="2012"/> </front> </reference> <reference anchor="vorbis-trim" target="http://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-130000A.2"> <front> <title>The Vorbis I Specification, Appendix A: Embedding Vorbis into an Ogg stream</title> <author initials="C." surname="Montgomery" fullname="Christopher "Monty" Montgomery"/> <date month="November" year="2008"/> </front> </reference> </references> </back> </rfc>