0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'O' | 'p' | 'u' | 's' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 'H' | 'e' | 'a' | 'd' | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Version = 1 | Channel Count | Pre-skip | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Input Sample Rate (Hz) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Output Gain (Q7.8 in dB) | Mapping Family| | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : | | : Optional Channel Mapping Table... : | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 2: ID Header Packet The fields in the identification (ID) header have the following meaning: 1. Magic Signature: This is an 8-octet (64-bit) field that allows codec identification and is human readable. It contains, in order, the magic numbers: 0x4F 'O' 0x70 'p' 0x75 'u'
0x73 's' 0x48 'H' 0x65 'e' 0x61 'a' 0x64 'd' Starting with "Op" helps distinguish it from audio data packets, as this is an invalid TOC sequence. 2. Version (8 bits, unsigned): The version number MUST always be '1' for this version of the encapsulation specification. Implementations SHOULD treat streams where the upper four bits of the version number match that of a recognized specification as backwards compatible with that specification. That is, the version number can be split into "major" and "minor" version sub-fields, with changes to the minor sub-field (in the lower four bits) signaling compatible changes. For example, an implementation of this specification SHOULD accept any stream with a version number of '15' or less, and SHOULD assume any stream with a version number '16' or greater is incompatible. The initial version '1' was chosen to keep implementations from relying on this octet as a null terminator for the "OpusHead" string. 3. Output Channel Count 'C' (8 bits, unsigned): This is the number of output channels. This might be different than the number of encoded channels, which can change on a packet-by-packet basis. This value MUST NOT be zero. The maximum allowable value depends on the channel mapping family, and might be as large as 255. See Section 5.1.1 for details. 4. Pre-skip (16 bits, unsigned, little endian): This is the number of samples (at 48 kHz) to discard from the decoder output when starting playback, and also the number to subtract from a page's granule position to calculate its PCM sample position. When cropping the beginning of existing Ogg Opus streams, a pre-skip of at least 3,840 samples (80 ms) is RECOMMENDED to ensure complete convergence in the decoder.
5. Input Sample Rate (32 bits, unsigned, little endian): This is the sample rate of the original input (before encoding), in Hz. This field is _not_ the sample rate to use for playback of the encoded data. Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and 20 kHz. Each packet in the stream can have a different audio bandwidth. Regardless of the audio bandwidth, the reference decoder supports decoding any stream at a sample rate of 8, 12, 16, 24, or 48 kHz. The original sample rate of the audio passed to the encoder is not preserved by the lossy compression. An Ogg Opus player SHOULD select the playback sample rate according to the following procedure: 1. If the hardware supports 48 kHz playback, decode at 48 kHz. 2. Otherwise, if the hardware's highest available sample rate is a supported rate, decode at this sample rate. 3. Otherwise, if the hardware's highest available sample rate is less than 48 kHz, decode at the next higher Opus supported rate above the highest available hardware rate and resample. 4. Otherwise, decode at 48 kHz and resample. However, the 'input sample rate' field allows the muxer to pass the sample rate of the original input stream as metadata. This is useful when the user requires the output sample rate to match the input sample rate. For example, when not playing the output, an implementation writing PCM format samples to disk might choose to resample the audio back to the original input sample rate to reduce surprise to the user, who might reasonably expect to get back a file with the same sample rate. A value of zero indicates "unspecified". Muxers SHOULD write the actual input sample rate or zero, but implementations that do something with this field SHOULD take care to behave sanely if given crazy values (e.g., do not actually upsample the output to 10 MHz if requested). Implementations SHOULD support input sample rates between 8 kHz and 192 kHz (inclusive). Rates outside this range MAY be ignored by falling back to the default rate of 48 kHz instead.
6. Output Gain (16 bits, signed, little endian): This is a gain to be applied when decoding. It is 20*log10 of the factor by which to scale the decoder output to achieve the desired playback volume, stored in a 16-bit, signed, two's complement fixed-point value with 8 fractional bits (i.e., Q7.8 [Q-NOTATION]). To apply the gain, an implementation could use the following: sample *= pow(10, output_gain/(20.0*256)) where 'output_gain' is the raw 16-bit value from the header. Players and media frameworks SHOULD apply it by default. If a player chooses to apply any volume adjustment or gain modification, such as the R128_TRACK_GAIN (see Section 5.2), the adjustment MUST be applied in addition to this output gain in order to achieve playback at the normalized volume. A muxer SHOULD set this field to zero, and instead apply any gain prior to encoding, when this is possible and does not conflict with the user's wishes. A nonzero output gain indicates the gain was adjusted after encoding, or that a user wished to adjust the gain for playback while preserving the ability to recover the original signal amplitude. Although the output gain has enormous range (+/- 128 dB, enough to amplify inaudible sounds to the threshold of physical pain), most applications can only reasonably use a small portion of this range around zero. The large range serves in part to ensure that gain can always be losslessly transferred between OpusHead and R128 gain tags (see below) without saturating. 7. Channel Mapping Family (8 bits, unsigned): This octet indicates the order and semantic meaning of the output channels. Each currently specified value of this octet indicates a mapping family, which defines a set of allowed channel counts, and the ordered set of channel names for each allowed channel count. The details are described in Section 5.1.1. 8. Channel Mapping Table: This table defines the mapping from encoded streams to output channels. Its contents are specified in Section 5.1.1.
All fields in the ID headers are REQUIRED, except for 'channel mapping table', which MUST be omitted when the channel mapping family is 0, but is REQUIRED otherwise. Implementations SHOULD treat a stream as invalid if it contains an ID header that does not have enough data for these fields, even if it contain a valid 'magic signature'. Future versions of this specification, even backwards- compatible versions, might include additional fields in the ID header. If an ID header has a compatible major version, but a larger minor version, an implementation MUST NOT treat it as invalid for containing additional data not specified here, provided it still completes on the first page. Figure 3. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+ | Stream Count | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Coupled Count | Channel Mapping... : +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 3: Channel Mapping Table The fields in the channel mapping table have the following meaning: 1. Stream Count 'N' (8 bits, unsigned): This is the total number of streams encoded in each Ogg packet. This value is necessary to correctly parse the packed Opus packets inside an Ogg packet, as described in Section 3. This value MUST NOT be zero, as without at least one Opus packet with a valid TOC sequence, a demuxer cannot recover the duration of an Ogg packet. For channel mapping family 0, this value defaults to 1, and is not coded.
2. Coupled Stream Count 'M' (8 bits, unsigned): This is the number of streams whose decoders are to be configured to produce two channels (stereo). This MUST be no larger than the total number of streams, N. Each packet in an Opus stream has an internal channel count of 1 or 2, which can change from packet to packet. This is selected by the encoder depending on the bitrate and the audio being encoded. The original channel count of the audio passed to the encoder is not necessarily preserved by the lossy compression. Regardless of the internal channel count, any Opus stream can be decoded as mono (a single channel) or stereo (two channels) by appropriate initialization of the decoder. The 'coupled stream count' field indicates that the decoders for the first M Opus streams are to be initialized for stereo (two-channel) output, and the remaining (N - M) decoders are to be initialized for mono (a single channel) only. The total number of decoded channels, (M + N), MUST be no larger than 255, as there is no way to index more channels than that in the channel mapping. For channel mapping family 0, this value defaults to (C - 1) (i.e., 0 for mono and 1 for stereo), and is not coded. 3. Channel Mapping (8*C bits): This contains one octet per output channel, indicating which decoded channel is to be used for each one. Let 'index' be the value of this octet for a particular output channel. This value MUST either be smaller than (M + N) or be the special value 255. If 'index' is less than 2*M, the output MUST be taken from decoding stream ('index'/2) as stereo and selecting the left channel if 'index' is even, and the right channel if 'index' is odd. If 'index' is 2*M or larger, but less than 255, the output MUST be taken from decoding stream ('index' - M) as mono. If 'index' is 255, the corresponding output channel MUST contain pure silence. The number of output channels, C, is not constrained to match the number of decoded channels (M + N). A single index value MAY appear multiple times, i.e., the same decoded channel might be mapped to multiple output channels. Some decoded channels might not be assigned to any output channel, as well.
For channel mapping family 0, the first index defaults to 0, and if C == 2, the second index defaults to 1. Neither index is coded. After producing the output channels, the channel mapping family determines the semantic meaning of each one. There are three defined mapping families in this specification. RFC7587]. o 1 channel: monophonic (mono). o 2 channels: stereo (left, right). Special mapping: This channel mapping family also indicates that the content consists of a single Opus stream that is stereo if and only if C == 2, with stream index 0 mapped to output channel 0 (mono, or left channel) and stream index 1 mapped to output channel 1 (right channel) if stereo. When the 'channel mapping family' octet has this value, the channel mapping table MUST be omitted from the ID header packet.
o 6 channels: 5.1 surround (front left, front center, front right, rear left, rear right, LFE). o 7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE). o 8 channels: 7.1 surround (front left, front center, front right, side left, side right, rear left, rear right, LFE). This set of surround options and speaker location orderings is the same as those used by the Vorbis codec [VORBIS-MAPPING]. The ordering is different from the one used by the WAVE [WAVE-MULTICHANNEL] and Free Lossless Audio Codec (FLAC) [FLAC] formats, so correct ordering requires permutation of the output channels when decoding to or encoding from those formats. "LFE" here refers to a Low Frequency Effects channel, often mapped to a subwoofer with no particular spatial position. Implementations SHOULD identify "side" or "rear" speaker locations with "surround" and "back" as appropriate when interfacing with audio formats or systems that prefer that terminology.
Implementations MAY use the matrices in Figures 4 through 9 to implement downmixing from multichannel files using channel mapping family 1 (Section 126.96.36.199), which are known to give acceptable results for stereo. Matrices for 3 and 4 channels are normalized so each coefficient row sums to 1 to avoid clipping. For 5 or more channels, they are normalized to 2 as a compromise between clipping and dynamic range reduction. In these matrices the front-left and front-right channels are generally passed through directly. When a surround channel is split between both the left and right stereo channels, coefficients are chosen so their squares sum to 1, which helps preserve the perceived intensity. Rear channels are mixed more diffusely or attenuated to maintain focus on the front channels. L output = ( 0.585786 * left + 0.414214 * center ) R output = ( 0.414214 * center + 0.585786 * right ) Exact coefficient values are 1 and 1/sqrt(2), multiplied by 1/(1 + 1/sqrt(2)) for normalization. Figure 4: Stereo Downmix Matrix for the Linear Surround Channel Mapping / \ / \ / FL \ | L output | | 0.422650 0.000000 0.366025 0.211325 | | FR | | R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL | \ / \ / \ RR / Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by 1/(1 + sqrt(3)/2 + 1/2) for normalization. Figure 5: Stereo Downmix Matrix for the Quadraphonic Channel Mapping / FL \ / \ / \ | FC | | L | | 0.650802 0.460186 0.000000 0.563611 0.325401 | | FR | | R | = | 0.000000 0.460186 0.650802 0.325401 0.563611 | | RL | \ / \ / | RR | \ / Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2) for normalization. Figure 6: Stereo Downmix Matrix for the 5.0 Surround Mapping
/FL \ / \ / \ |FC | |L| | 0.529067 0.374107 0.000000 0.458186 0.264534 0.374107 | |FR | |R| = | 0.000000 0.374107 0.529067 0.264534 0.458186 0.374107 | |RL | \ / \ / |RR | \LFE/ Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2)) for normalization. Figure 7: Stereo Downmix Matrix for the 5.1 Surround Mapping / \ | 0.455310 0.321953 0.000000 0.394310 0.227655 0.278819 0.321953 | | 0.000000 0.321953 0.455310 0.227655 0.394310 0.278819 0.321953 | \ / Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and sqrt(3)/2/sqrt(2), multiplied by 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization. The coefficients are in the same order as in Section 188.8.131.52 and the matrices above. Figure 8: Stereo Downmix Matrix for the 6.1 Surround Mapping / \ | .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 | | .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 | \ / Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 2/(2 + 2/sqrt(2) + sqrt(3)) for normalization. The coefficients are in the same order as in Section 184.108.40.206 and the matrices above. Figure 9: Stereo Downmix Matrix for the 7.1 Surround Mapping
Figure 10: Comment Header Packet The comment header consists of a 64-bit 'magic signature' field, followed by data in the same format as the [VORBIS-COMMENT] header used in Ogg Vorbis, except (like Ogg Theora and Speex) the final 'framing bit' specified in the Vorbis specification is not present.
1. Magic Signature: This is an 8-octet (64-bit) field that allows codec identification and is human readable. It contains, in order, the magic numbers: 0x4F 'O' 0x70 'p' 0x75 'u' 0x73 's' 0x54 'T' 0x61 'a' 0x67 'g' 0x73 's' Starting with "Op" helps distinguish it from audio data packets, as this is an invalid TOC sequence. 2. Vendor String Length (32 bits, unsigned, little endian): This field gives the length of the following vendor string, in octets. It MUST NOT indicate that the vendor string is longer than the rest of the packet. 3. Vendor String (variable length, UTF-8 vector): This is a simple human-readable tag for vendor information, encoded as a UTF-8 string [RFC3629]. No terminating null octet is necessary. This tag is intended to identify the codec encoder and encapsulation implementations, for tracing differences in technical behavior. User-facing applications can use the 'ENCODER' user comment tag to identify themselves.
4. User Comment List Length (32 bits, unsigned, little endian): This field indicates the number of user-supplied comments. It MAY indicate there are zero user-supplied comments, in which case there are no additional fields in the packet. It MUST NOT indicate that there are so many comments that the comment string lengths would require more data than is available in the rest of the packet. 5. User Comment #i String Length (32 bits, unsigned, little endian): This field gives the length of the following user comment string, in octets. There is one for each user comment indicated by the 'user comment list length' field. It MUST NOT indicate that the string is longer than the rest of the packet. 6. User Comment #i String (variable length, UTF-8 vector): This field contains a single user comment encoded as a UTF-8 string [RFC3629]. There is one for each user comment indicated by the 'user comment list length' field. The 'vendor string length' and 'user comment list length' fields are REQUIRED, and implementations SHOULD treat a stream as invalid if it contains a comment header that does not have enough data for these fields, or that does not contain enough data for the corresponding vendor string or user comments they describe. Making this check before allocating the associated memory to contain the data helps prevent a possible Denial-of-Service (DoS) attack from small comment headers that claim to contain strings longer than the entire packet or more user comments than could possibly fit in the packet. Immediately following the user comment list, the comment header MAY contain zero-padding or other binary data that is not specified here. If the least-significant bit of the first byte of this data is 1, then editors SHOULD preserve the contents of this data when updating the tags, but if this bit is 0, all such data MAY be treated as padding, and truncated or discarded as desired. This allows informal experimentation with the format of this binary data until it can be specified later. The comment header can be arbitrarily large and might be spread over a large number of Ogg pages. Implementations MUST avoid attempting to allocate excessive amounts of memory when presented with a very large comment header. To accomplish this, implementations MAY treat a stream as invalid if it has a comment header larger than
125,829,120 octets (120 MB), and MAY ignore individual comments that are not fully contained within the first 61,440 octets of the comment header. VORBIS-COMMENT] with the same recommended tag names: ARTIST, TITLE, DATE, ALBUM, and so on. Two new comment tags are introduced here: First, an optional gain for track normalization: R128_TRACK_GAIN=-573 representing the volume shift needed to normalize the track's volume during isolated playback, in random shuffle, and so on. The gain is a Q7.8 fixed-point number in dB, as in the ID header's 'output gain' field. This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in Vorbis [REPLAY-GAIN], except that the normal volume reference is the [EBU-R128] standard. Second, an optional gain for album normalization: R128_ALBUM_GAIN=111 representing the volume shift needed to normalize the overall volume when played as part of a particular collection of tracks. The gain is also a Q7.8 fixed-point number in dB, as in the ID header's 'output gain' field. The values '-573' and '111' given here are just examples. An Ogg Opus stream MUST NOT have more than one of each of these tags, and, if present, their values MUST be an integer from -32768 to 32767, inclusive, represented in ASCII as a base 10 number with no whitespace. A leading '+' or '-' character is valid. Leading zeros are also permitted, but the value MUST be represented by no more than 6 characters. Other non-digit characters MUST NOT be present. If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent the R128 normalization gain relative to the 'output gain' field specified in the ID header. If a player chooses to make use of the R128_TRACK_GAIN tag or the R128_ALBUM_GAIN tag, it MUST apply those gains _in addition_ to the 'output gain' value. If a tool modifies the ID header's 'output gain' field, it MUST also update or
remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present. A muxer SHOULD place the gain it wants other tools to use by default into the 'output gain' field, and not the comment tag. To avoid confusion with multiple normalization schemes, an Opus comment header SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK, REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags, unless they are only to be used in some context where there is guaranteed to be no such confusion. [EBU-R128] normalization is preferred to the earlier REPLAYGAIN schemes because of its clear definition and adoption by industry. Peak normalizations are difficult to calculate reliably for lossy codecs because of variation in excursion heights due to decoder differences. In the authors' investigations, they were not applied consistently or broadly enough to merit inclusion here.
efficient manner allowed (a VBR code 3 Opus packet). Even in such a packet, most of the data will be zeros as 2.5 ms frames cannot actually use all 1275 octets. The largest packet consisting of entirely useful data is (15,326*N - 2) octets. This corresponds to 120 ms of audio encoded as 10 ms frames in either SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little sense for the quality achieved. A more reasonable limit is (7,664*N - 2) octets. This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode frames, with a total bitrate just under 511 kbps (not counting the Ogg encapsulation overhead). For channel mapping family 1, N = 8 provides a reasonable upper bound, as it allows for each of the 8 possible output channels to be decoded from a separate stereo Opus stream. This gives a size of 61,310 octets, which is rounded up to a multiple of 1,024 octets to yield the audio data packet size of 61,440 octets that any implementation is expected to be able to process successfully. RFC6716], the number of samples can be queried with opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples)); To achieve good quality in the very first samples of a stream, implementations MAY use linear predictive coding (LPC) extrapolation to generate at least 120 extra samples at the beginning to avoid the Opus encoder having to encode a discontinuous signal. For more information on linear prediction, see [LINEAR-PREDICTION]. For an input file containing 'length' samples, the implementation SHOULD set the 'pre-skip' header value to (delay_samples + extra_samples), encode at least (length + delay_samples + extra_samples) samples, and set the granule position of the last page to (length + delay_samples + extra_samples). This ensures that the encoded file has the same duration as the original, with no time offset. The best way to pad the end of the stream is to also use LPC extrapolation, but zero-padding is also acceptable.
LPC-SAMPLE]. When extending the end of the signal, order-N (typically with N ranging from 8 to 40) LPC analysis is performed on a window near the end of the signal. The last N samples are used as memory to an infinite impulse response (IIR) filter. The filter is then applied on a zero input to extrapolate the end of the signal. Let 'a(k)' be the kth LPC coefficient and 'x(n)' be the nth sample of the signal. Each new sample past the end of the signal is computed as N --- x(n) = \ a(k)*x(n - k) / --- k = 1 The process is repeated independently for each channel. It is possible to extend the beginning of the signal by applying the same process backward in time. When extending the beginning of the signal, it is best to apply a "fade in" to the extrapolated signal, e.g., by multiplying it by a half-Hanning window [HANNING].
5. Continue the encoding process normally from there, without any reset to the encoder. In encoders derived from the reference implementation, inter-frame prediction can be turned off by calling opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1)); For best results, this implementation requires that prediction be explicitly enabled again before resuming normal encoding, even after a reset. RFC4732]. This is just as much a problem for the container as it is for the codec itself. Malicious payloads and/or input streams can be used to attack codec implementations. Implementations MUST NOT overrun their allocated memory nor consume excessive resources when decoding payloads or processing input streams. Although problems in encoding applications are typically rarer, this still applies to a muxer, as vulnerabilities would allow an attacker to attack transcoding gateways. Header parsing code contains the most likely area for potential overruns. It is important for implementations to ensure their buffers contain enough data for all of the required fields before attempting to read it (for example, for all of the channel map data in the ID header). Implementations would do well to validate the indices of the channel map, also, to ensure they meet all of the restrictions outlined in Section 5.1.1, in order to avoid attempting to read data from channels that do not exist. To avoid excessive resource usage, we advise implementations to be especially wary of streams that might cause them to process far more data than was actually transmitted. For example, a relatively small comment header may contain values for the string lengths or user comment list length that imply that it is many gigabytes in size. Even computing the size of the required buffer could overflow a 32-bit integer, and actually attempting to allocate such a buffer before verifying it would be a reasonable size is a bad idea. After reading the user comment list length, implementations might wish to verify that the header contains at least the minimum amount of data for that many comments (4 additional octets per comment, to indicate each has a length of zero) before proceeding any further, again taking care to avoid overflow in these calculations. If allocating
an array of pointers to point at these strings, the size of the pointers may be larger than 4 octets, potentially requiring a separate overflow check. Another bug in this class we have observed more than once involves the handling of invalid data at the end of a stream. Often, implementations will seek to the end of a stream to locate the last timestamp in order to compute its total duration. If they do not find a valid capture pattern and Ogg page from the desired logical stream, they will back up and try again. If care is not taken to avoid re-scanning data that was already scanned, this search can quickly devolve into something with a complexity that is quadratic in the amount of invalid data. In general, when seeking, implementations will wish to be cautious about the effects of invalid granule position values and ensure all algorithms will continue to make progress and eventually terminate, even if these are missing or out of order. Like most other container formats, Ogg Opus streams SHOULD NOT be used with insecure ciphers or cipher modes that are vulnerable to known-plaintext attacks. Elements such as the Ogg page capture pattern and the 'magic signature' fields in the ID header and the comment header all have easily predictable values, in addition to various elements of the codec data itself. RFC6381] and [RFC5334], e.g., audio/ogg; codecs=opus for an Ogg Opus file. The RECOMMENDED filename extension for Ogg Opus files is '.opus'. When Opus is concurrently multiplexed with other streams in an Ogg container, one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg" mime-types, as defined in [RFC5334]. Such streams are not strictly "Ogg Opus files" as described above, since they
contain more than a single Opus stream per sequentially multiplexed segment, e.g., video or multiple audio tracks. In such cases, the '.opus' filename extension is NOT RECOMMENDED. In either case, this document updates [RFC5334] to add "opus" as a codecs parameter value with char: 'OpusHead' as Codec Identifier. RFC5334] for "audio/ogg", "video/ogg", and "application/ogg" Media Types. This document defines a new registry "Opus Channel Mapping Families" to indicate how the semantic meanings of the channels in a multi- channel Opus stream are described. IANA has created a new namespace of "Opus Channel Mapping Families". This registry is listed on the IANA Matrix. Modifications to this registry follow the "Specification Required" registration policy as defined in [RFC5226]. Each registry entry consists of a Channel Mapping Family Number, which is specified in decimal in the range 0 to 255, inclusive, and a Reference (or list of references). Each Reference must point to sufficient documentation to describe what information is coded in the Opus identification header for this channel mapping family, how a demuxer determines the stream count ('N') and coupled stream count ('M') from this information, and how it determines the proper interpretation of each of the decoded channels. This document defines three initial assignments for this registry. +-------+---------------------------+ | Value | Reference | +-------+---------------------------+ | 0 | RFC 7845, Section 220.127.116.11 | | | | | 1 | RFC 7845, Section 18.104.22.168 | | | | | 255 | RFC 7845, Section 22.214.171.124 | +-------+---------------------------+ The designated expert will determine if the Reference points to a specification that meets the requirements for permanence and ready availability laid out in [RFC5226] and whether it specifies the information described above with sufficient clarity to allow interoperable implementations.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <https://www.rfc-editor.org/info/rfc2119>. [RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0", RFC 3533, DOI 10.17487/RFC3533, May 2003, <https://www.rfc-editor.org/info/rfc3533>. [RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November 2003, <https://www.rfc-editor.org/info/rfc3629>. [RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 5226, DOI 10.17487/RFC5226, May 2008, <https://www.rfc-editor.org/info/rfc5226>. [RFC5334] Goncalves, I., Pfeiffer, S., and C. Montgomery, "Ogg Media Types", RFC 5334, DOI 10.17487/RFC5334, September 2008, <https://www.rfc-editor.org/info/rfc5334>. [RFC6381] Gellens, R., Singer, D., and P. Frojdh, "The 'Codecs' and 'Profiles' Parameters for "Bucket" Media Types", RFC 6381, DOI 10.17487/RFC6381, August 2011, <https://www.rfc-editor.org/info/rfc6381>. [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716, September 2012, <https://www.rfc-editor.org/info/rfc6716>. [EBU-R128] EBU Technical Committee, "Loudness Recommendation EBU R128", August 2011, <https://tech.ebu.ch/loudness>. [VORBIS-COMMENT] Montgomery, C., "Ogg Vorbis I Format Specification: Comment Field and Header Specification", July 2002, <https://www.xiph.org/vorbis/doc/v-comment.html>.
[RFC4732] Handley, M., Ed., Rescorla, E., Ed., and IAB, "Internet Denial-of-Service Considerations", RFC 4732, DOI 10.17487/RFC4732, December 2006, <https://www.rfc-editor.org/info/rfc4732>. [RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format for the Opus Speech and Audio Codec", RFC 7587, DOI 10.17487/RFC7587, June 2015, <https://www.rfc-editor.org/info/rfc7587>. [FLAC] Coalson, J., "FLAC - Free Lossless Audio Codec Format Description", January 2008, <https://xiph.org/flac/format.html>. [HANNING] Wikipedia, "Hann window", February 2016, <https://en.wikipedia.org/w/index.php?title=Window_functio n&oldid=703074467#Hann_.28Hanning.29_window>. [LINEAR-PREDICTION] Wikipedia, "Linear Predictive Coding", October 2015, <https://en.wikipedia.org/w/ index.php?title=Linear_predictive_coding&oldid=687498962>. [LPC-SAMPLE] Degener, J. and C. Bormann, "Autocorrelation LPC coeff generation algorithm (Vorbis source code)", November 1994, <https://svn.xiph.org/trunk/vorbis/lib/lpc.c>. [Q-NOTATION] Wikipedia, "Q (number format)", December 2015, <https://en.wikipedia.org/w/ index.php?title=Q_%28number_format%29&oldid=697252615>. [REPLAY-GAIN] Parker, C. and M. Leese, "VorbisComment: Replay Gain", June 2009, <https://wiki.xiph.org/VorbisComment#Replay_Gain>. [SEEKING] Pfeiffer, S., Parker, C., and G. Maxwell, "Granulepos Encoding and How Seeking Really Works", May 2012, <https://wiki.xiph.org/Seeking>.
[VORBIS-MAPPING] Montgomery, C., "The Vorbis I Specification, Section 4.3.9 Output Channel Order", January 2010, <https://www.xiph.org/vorbis/doc/ Vorbis_I_spec.html#x1-810004.3.9>. [VORBIS-TRIM] Montgomery, C., "The Vorbis I Specification, Appendix A: Embedding Vorbis into an Ogg stream", November 2008, <https://xiph.org/vorbis/doc/ Vorbis_I_spec.html#x1-132000A.2>. [WAVE-MULTICHANNEL] Microsoft Corporation, "Multiple Channel Audio Data and WAVE Files", March 2007, <https://msdn.microsoft.com/en-us/windows/hardware/ gg463006.aspx>.