1 <?xml version="1.0" encoding="utf-8"?> 2 <!-- 3 Copyright (c) 2012-2016 Xiph.Org Foundation and contributors 4 5 Redistribution and use in source and binary forms, with or without 6 modification, are permitted provided that the following conditions 7 are met: 8 9 - Redistributions of source code must retain the above copyright 10 notice, this list of conditions and the following disclaimer. 11 12 - Redistributions in binary form must reproduce the above copyright 13 notice, this list of conditions and the following disclaimer in the 14 documentation and/or other materials provided with the distribution. 15 16 THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS 17 ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT 18 LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR 19 A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER 20 OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, 21 EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 22 PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR 23 PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF 24 LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING 25 NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS 26 SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 27 28 Special permission is granted to remove the above copyright notice, list of 29 conditions, and disclaimer when submitting this document, with or without 30 modification, to the IETF. 31 --> 32 <!DOCTYPE rfc SYSTEM 'rfc2629.dtd' [ 33 <!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'> 34 <!ENTITY rfc3533 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3533.xml'> 35 <!ENTITY rfc3629 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3629.xml'> 36 <!ENTITY rfc4732 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4732.xml'> 37 <!ENTITY rfc5226 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5226.xml'> 38 <!ENTITY rfc5334 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5334.xml'> 39 <!ENTITY rfc6381 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6381.xml'> 40 <!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'> 41 <!ENTITY rfc6982 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6982.xml'> 42 <!ENTITY rfc7587 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.7587.xml'> 43 ]> 44 <?rfc toc="yes" symrefs="yes" ?> 45 46 <rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-14" 47 updates="5334"> 48 49 <front> 50 <title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title> 51 <author initials="T.B." surname="Terriberry" fullname="Timothy B. Terriberry"> 52 <organization>Mozilla Corporation</organization> 53 <address> 54 <postal> 55 <street>650 Castro Street</street> 56 <city>Mountain View</city> 57 <region>CA</region> 58 <code>94041</code> 59 <country>USA</country> 60 </postal> 61 <phone>+1 650 903-0800</phone> 62 <email>tterribe (a] xiph.org</email> 63 </address> 64 </author> 65 66 <author initials="R." surname="Lee" fullname="Ron Lee"> 67 <organization>Voicetronix</organization> 68 <address> 69 <postal> 70 <street>246 Pulteney Street, Level 1</street> 71 <city>Adelaide</city> 72 <region>SA</region> 73 <code>5000</code> 74 <country>Australia</country> 75 </postal> 76 <phone>+61 8 8232 9112</phone> 77 <email>ron (a] debian.org</email> 78 </address> 79 </author> 80 81 <author initials="R." surname="Giles" fullname="Ralph Giles"> 82 <organization>Mozilla Corporation</organization> 83 <address> 84 <postal> 85 <street>163 West Hastings Street</street> 86 <city>Vancouver</city> 87 <region>BC</region> 88 <code>V6B 1H5</code> 89 <country>Canada</country> 90 </postal> 91 <phone>+1 778 785 1540</phone> 92 <email>giles (a] xiph.org</email> 93 </address> 94 </author> 95 96 <date day="22" month="February" year="2016"/> 97 <area>RAI</area> 98 <workgroup>codec</workgroup> 99 100 <abstract> 101 <t> 102 This document defines the Ogg encapsulation for the Opus interactive speech and 103 audio codec. 104 This allows data encoded in the Opus format to be stored in an Ogg logical 105 bitstream. 106 </t> 107 </abstract> 108 </front> 109 110 <middle> 111 <section anchor="intro" title="Introduction"> 112 <t> 113 The IETF Opus codec is a low-latency audio codec optimized for both voice and 114 general-purpose audio. 115 See <xref target="RFC6716"/> for technical details. 116 This document defines the encapsulation of Opus in a continuous, logical Ogg 117 bitstream <xref target="RFC3533"/>. 118 Ogg encapsulation provides Opus with a long-term storage format supporting 119 all of the essential features, including metadata, fast and accurate seeking, 120 corruption detection, recapture after errors, low overhead, and the ability to 121 multiplex Opus with other codecs (including video) with minimal buffering. 122 It also provides a live streamable format, capable of delivery over a reliable 123 stream-oriented transport, without requiring all the data, or even the total 124 length of the data, up-front, in a form that is identical to the on-disk 125 storage format. 126 </t> 127 <t> 128 Ogg bitstreams are made up of a series of 'pages', each of which contains data 129 from one or more 'packets'. 130 Pages are the fundamental unit of multiplexing in an Ogg stream. 131 Each page is associated with a particular logical stream and contains a capture 132 pattern and checksum, flags to mark the beginning and end of the logical 133 stream, and a 'granule position' that represents an absolute position in the 134 stream, to aid seeking. 135 A single page can contain up to 65,025 octets of packet data from up to 255 136 different packets. 137 Packets can be split arbitrarily across pages, and continued from one page to 138 the next (allowing packets much larger than would fit on a single page). 139 Each page contains 'lacing values' that indicate how the data is partitioned 140 into packets, allowing a demultiplexer (demuxer) to recover the packet 141 boundaries without examining the encoded data. 142 A packet is said to 'complete' on a page when the page contains the final 143 lacing value corresponding to that packet. 144 </t> 145 <t> 146 This encapsulation defines the contents of the packet data, including 147 the necessary headers, the organization of those packets into a logical 148 stream, and the interpretation of the codec-specific granule position field. 149 It does not attempt to describe or specify the existing Ogg container format. 150 Readers unfamiliar with the basic concepts mentioned above are encouraged to 151 review the details in <xref target="RFC3533"/>. 152 </t> 153 154 </section> 155 156 <section anchor="terminology" title="Terminology"> 157 <t> 158 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", 159 "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this 160 document are to be interpreted as described in <xref target="RFC2119"/>. 161 </t> 162 163 </section> 164 165 <section anchor="packet_organization" title="Packet Organization"> 166 <t> 167 An Ogg Opus stream is organized as follows (see 168 <xref target="packet-org-example"/> for an example). 169 </t> 170 171 <figure anchor="packet-org-example" 172 title="Example packet organization for a logical Ogg Opus stream" 173 align="center"> 174 <artwork align="center"><![CDATA[ 175 Page 0 Pages 1 ... n Pages (n+1) ... 176 +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +-- 177 | | | | | | | | | | | | | 178 |+----------+| |+-----------------+| |+-------------------+ +----- 179 |||ID Header|| || Comment Header || ||Audio Data Packet 1| | ... 180 |+----------+| |+-----------------+| |+-------------------+ +----- 181 | | | | | | | | | | | | | 182 +------------+ +---+ +---+ ... +---+ +-----------+ +---------+ +-- 183 ^ ^ ^ 184 | | | 185 | | Mandatory Page Break 186 | | 187 | ID header is contained on a single page 188 | 189 'Beginning Of Stream' 190 ]]></artwork> 191 </figure> 192 193 <t> 194 There are two mandatory header packets. 195 The first packet in the logical Ogg bitstream MUST contain the identification 196 (ID) header, which uniquely identifies a stream as Opus audio. 197 The format of this header is defined in <xref target="id_header"/>. 198 It is placed alone (without any other packet data) on the first page of 199 the logical Ogg bitstream, and completes on that page. 200 This page has its 'beginning of stream' flag set. 201 </t> 202 <t> 203 The second packet in the logical Ogg bitstream MUST contain the comment header, 204 which contains user-supplied metadata. 205 The format of this header is defined in <xref target="comment_header"/>. 206 It MAY span multiple pages, beginning on the second page of the logical 207 stream. 208 However many pages it spans, the comment header packet MUST finish the page on 209 which it completes. 210 </t> 211 <t> 212 All subsequent pages are audio data pages, and the Ogg packets they contain are 213 audio data packets. 214 Each audio data packet contains one Opus packet for each of N different 215 streams, where N is typically one for mono or stereo, but MAY be greater than 216 one for multichannel audio. 217 The value N is specified in the ID header (see 218 <xref target="channel_mapping"/>), and is fixed over the entire length of the 219 logical Ogg bitstream. 220 </t> 221 <t> 222 The first (N - 1) Opus packets, if any, are packed one after another 223 into the Ogg packet, using the self-delimiting framing from Appendix B of 224 <xref target="RFC6716"/>. 225 The remaining Opus packet is packed at the end of the Ogg packet using the 226 regular, undelimited framing from Section 3 of <xref target="RFC6716"/>. 227 All of the Opus packets in a single Ogg packet MUST be constrained to have the 228 same duration. 229 An implementation of this specification SHOULD treat any Opus packet whose 230 duration is different from that of the first Opus packet in an Ogg packet as 231 if it were a malformed Opus packet with an invalid Table Of Contents (TOC) 232 sequence. 233 </t> 234 <t> 235 The TOC sequence at the beginning of each Opus packet indicates the coding 236 mode, audio bandwidth, channel count, duration (frame size), and number of 237 frames per packet, as described in Section 3.1 238 of <xref target="RFC6716"/>. 239 The coding mode is one of SILK, Hybrid, or Constrained Energy Lapped Transform 240 (CELT). 241 The combination of coding mode, audio bandwidth, and frame size is referred to 242 as the configuration of an Opus packet. 243 </t> 244 <t> 245 Packets are placed into Ogg pages in order until the end of stream. 246 Audio data packets might span page boundaries. 247 The first audio data page could have the 'continued packet' flag set 248 (indicating the first audio data packet is continued from a previous page) if, 249 for example, it was a live stream joined mid-broadcast, with the headers 250 pasted on the front. 251 If a page has the 'continued packet' flag set and one of the following 252 conditions is also true: 253 <list style="symbols"> 254 <t>the previous page with packet data does not end in a continued packet (does 255 not end with a lacing value of 255) OR</t> 256 <t>the page sequence numbers are not consecutive,</t> 257 </list> 258 then a demuxer MUST NOT attempt to decode the data for the first packet on the 259 page unless the demuxer has some special knowledge that would allow it to 260 interpret this data despite the missing pieces. 261 An implementation MUST treat a zero-octet audio data packet as if it were a 262 malformed Opus packet as described in 263 Section 3.4 of <xref target="RFC6716"/>. 264 </t> 265 <t> 266 A logical stream ends with a page with the 'end of stream' flag set, but 267 implementations need to be prepared to deal with truncated streams that do not 268 have a page marked 'end of stream'. 269 There is no reason for the final packet on the last page to be a continued 270 packet, i.e., for the final lacing value to be 255. 271 However, demuxers might encounter such streams, possibly as the result of a 272 transfer that did not complete or of corruption. 273 If a packet continues onto a subsequent page (i.e., when the page ends with a 274 lacing value of 255) and one of the following conditions is also true: 275 <list style="symbols"> 276 <t>the next page with packet data does not have the 'continued packet' flag 277 set OR</t> 278 <t>there is no next page with packet data OR</t> 279 <t>the page sequence numbers are not consecutive,</t> 280 </list> 281 then a demuxer MUST NOT attempt to decode the data from that packet unless the 282 demuxer has some special knowledge that would allow it to interpret this data 283 despite the missing pieces. 284 There MUST NOT be any more pages in an Opus logical bitstream after a page 285 marked 'end of stream'. 286 </t> 287 </section> 288 289 <section anchor="granpos" title="Granule Position"> 290 <t> 291 The granule position MUST be zero for the ID header page and the 292 page where the comment header completes. 293 That is, the first page in the logical stream, and the last header 294 page before the first audio data page both have a granule position of zero. 295 </t> 296 <t> 297 The granule position of an audio data page encodes the total number of PCM 298 samples in the stream up to and including the last fully-decodable sample from 299 the last packet completed on that page. 300 The granule position of the first audio data page will usually be larger than 301 zero, as described in <xref target="start_granpos_restrictions"/>. 302 </t> 303 304 <t> 305 A page that is entirely spanned by a single packet (that completes on a 306 subsequent page) has no granule position, and the granule position field is 307 set to the special value '-1' in two's complement. 308 </t> 309 310 <t> 311 The granule position of an audio data page is in units of PCM audio samples at 312 a fixed rate of 48 kHz (per channel; a stereo stream's granule position 313 does not increment at twice the speed of a mono stream). 314 It is possible to run an Opus decoder at other sampling rates, 315 but all Opus packets encode samples at a sampling rate that evenly divides 316 48 kHz. 317 Therefore, the value in the granule position field always counts samples 318 assuming a 48 kHz decoding rate, and the rest of this specification makes 319 the same assumption. 320 </t> 321 322 <t> 323 The duration of an Opus packet as defined in <xref target="RFC6716"/> can be 324 any multiple of 2.5 ms, up to a maximum of 120 ms. 325 This duration is encoded in the TOC sequence at the beginning of each packet. 326 The number of samples returned by a decoder corresponds to this duration 327 exactly, even for the first few packets. 328 For example, a 20 ms packet fed to a decoder running at 48 kHz will 329 always return 960 samples. 330 A demuxer can parse the TOC sequence at the beginning of each Ogg packet to 331 work backwards or forwards from a packet with a known granule position (i.e., 332 the last packet completed on some page) in order to assign granule positions 333 to every packet, or even every individual sample. 334 The one exception is the last page in the stream, as described below. 335 </t> 336 337 <t> 338 All other pages with completed packets after the first MUST have a granule 339 position equal to the number of samples contained in packets that complete on 340 that page plus the granule position of the most recent page with completed 341 packets. 342 This guarantees that a demuxer can assign individual packets the same granule 343 position when working forwards as when working backwards. 344 For this to work, there cannot be any gaps. 345 </t> 346 347 <section anchor="gap-repair" title="Repairing Gaps in Real-time Streams"> 348 <t> 349 In order to support capturing a real-time stream that has lost or not 350 transmitted packets, a multiplexer (muxer) SHOULD emit packets that explicitly 351 request the use of Packet Loss Concealment (PLC) in place of the missing 352 packets. 353 Implementations that fail to do so still MUST NOT increment the granule 354 position for a page by anything other than the number of samples contained in 355 packets that actually complete on that page. 356 </t> 357 <t> 358 Only gaps that are a multiple of 2.5 ms are repairable, as these are the 359 only durations that can be created by packet loss or discontinuous 360 transmission. 361 Muxers need not handle other gap sizes. 362 Creating the necessary packets involves synthesizing a TOC byte (defined in 363 Section 3.1 of <xref target="RFC6716"/>)—and whatever 364 additional internal framing is needed—to indicate the packet duration 365 for each stream. 366 The actual length of each missing Opus frame inside the packet is zero bytes, 367 as defined in Section 3.2.1 of <xref target="RFC6716"/>. 368 </t> 369 370 <t> 371 Zero-byte frames MAY be packed into packets using any of codes 0, 1, 372 2, or 3. 373 When successive frames have the same configuration, the higher code packings 374 reduce overhead. 375 Likewise, if the TOC configuration matches, the muxer MAY further combine the 376 empty frames with previous or subsequent non-zero-length frames (using 377 code 2 or VBR code 3). 378 </t> 379 380 <t> 381 <xref target="RFC6716"/> does not impose any requirements on the PLC, but this 382 section outlines choices that are expected to have a positive influence on 383 most PLC implementations, including the reference implementation. 384 Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth, 385 channel count, and frame size as the previous packet (if any). 386 This is the simplest and usually the most well-tested case for the PLC to 387 handle and it covers all losses that do not include a configuration switch, 388 as defined in Section 4.5 of <xref target="RFC6716"/>. 389 </t> 390 391 <t> 392 When a previous packet is available, keeping the audio bandwidth and channel 393 count the same allows the PLC to provide maximum continuity in the concealment 394 data it generates. 395 However, if the size of the gap is not a multiple of the most recent frame 396 size, then the frame size will have to change for at least some frames. 397 Such changes SHOULD be delayed as long as possible to simplify 398 things for PLC implementations. 399 </t> 400 401 <t> 402 As an example, a 95 ms gap could be encoded as nineteen 5 ms frames 403 in two bytes with a single CBR code 3 packet. 404 If the previous frame size was 20 ms, using four 20 ms frames 405 followed by three 5 ms frames requires 4 bytes (plus an extra byte 406 of Ogg lacing overhead), but allows the PLC to use its well-tested steady 407 state behavior for as long as possible. 408 The total bitrate of the latter approach, including Ogg overhead, is about 409 0.4 kbps, so the impact on file size is minimal. 410 </t> 411 412 <t> 413 Changing modes is discouraged, since this causes some decoder implementations 414 to reset their PLC state. 415 However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple 416 of 10 ms. 417 If switching to CELT mode is needed to match the gap size, a muxer SHOULD do 418 so at the end of the gap to allow the PLC to function for as long as possible. 419 </t> 420 421 <t> 422 In the example above, if the previous frame was a 20 ms SILK mode frame, 423 the better solution is to synthesize a packet describing four 20 ms SILK 424 frames, followed by a packet with a single 10 ms SILK 425 frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms 426 gap. 427 This also requires four bytes to describe the synthesized packet data (two 428 bytes for a CBR code 3 and one byte each for two code 0 packets) but three 429 bytes of Ogg lacing overhead are needed to mark the packet boundaries. 430 At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality 431 solution. 432 </t> 433 434 <t> 435 Since medium-band audio is an option only in the SILK mode, wideband frames 436 SHOULD be generated if switching from that configuration to CELT mode, to 437 ensure that any PLC implementation which does try to migrate state between 438 the modes will be able to preserve all of the available audio bandwidth. 439 </t> 440 441 </section> 442 443 <section anchor="preskip" title="Pre-skip"> 444 <t> 445 There is some amount of latency introduced during the decoding process, to 446 allow for overlap in the CELT mode, stereo mixing in the SILK mode, and 447 resampling. 448 The encoder might have introduced additional latency through its own resampling 449 and analysis (though the exact amount is not specified). 450 Therefore, the first few samples produced by the decoder do not correspond to 451 real input audio, but are instead composed of padding inserted by the encoder 452 to compensate for this latency. 453 These samples need to be stored and decoded, as Opus is an asymptotically 454 convergent predictive codec, meaning the decoded contents of each frame depend 455 on the recent history of decoder inputs. 456 However, a player will want to skip these samples after decoding them. 457 </t> 458 459 <t> 460 A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals 461 the number of samples that SHOULD be skipped (decoded but discarded) at the 462 beginning of the stream, though some specific applications might have a reason 463 for looking at that data. 464 This amount need not be a multiple of 2.5 ms, MAY be smaller than a single 465 packet, or MAY span the contents of several packets. 466 These samples are not valid audio. 467 </t> 468 469 <t> 470 For example, if the first Opus frame uses the CELT mode, it will always 471 produce 120 samples of windowed overlap-add data. 472 However, the overlap data is initially all zeros (since there is no prior 473 frame), meaning this cannot, in general, accurately represent the original 474 audio. 475 The SILK mode requires additional delay to account for its analysis and 476 resampling latency. 477 The encoder delays the original audio to avoid this problem. 478 </t> 479 480 <t> 481 The pre-skip field MAY also be used to perform sample-accurate cropping of 482 already encoded streams. 483 In this case, a value of at least 3840 samples (80 ms) provides 484 sufficient history to the decoder that it will have converged 485 before the stream's output begins. 486 </t> 487 488 </section> 489 490 <section anchor="pcm_sample_position" title="PCM Sample Position"> 491 <t> 492 The PCM sample position is determined from the granule position using the 493 formula 494 </t> 495 <figure align="center"> 496 <artwork align="center"><![CDATA[ 497 'PCM sample position' = 'granule position' - 'pre-skip' . 498 ]]></artwork> 499 </figure> 500 501 <t> 502 For example, if the granule position of the first audio data page is 59,971, 503 and the pre-skip is 11,971, then the PCM sample position of the last decoded 504 sample from that page is 48,000. 505 </t> 506 <t> 507 This can be converted into a playback time using the formula 508 </t> 509 <figure align="center"> 510 <artwork align="center"><![CDATA[ 511 'PCM sample position' 512 'playback time' = --------------------- . 513 48000.0 514 ]]></artwork> 515 </figure> 516 517 <t> 518 The initial PCM sample position before any samples are played is normally '0'. 519 In this case, the PCM sample position of the first audio sample to be played 520 starts at '1', because it marks the time on the clock 521 <spanx style="emph">after</spanx> that sample has been played, and a stream 522 that is exactly one second long has a final PCM sample position of '48000', 523 as in the example here. 524 </t> 525 526 <t> 527 Vorbis streams use a granule position smaller than the number of audio samples 528 contained in the first audio data page to indicate that some of those samples 529 are trimmed from the output (see <xref target="vorbis-trim"/>). 530 However, to do so, Vorbis requires that the first audio data page contains 531 exactly two packets, in order to allow the decoder to perform PCM position 532 adjustments before needing to return any PCM data. 533 Opus uses the pre-skip mechanism for this purpose instead, since the encoder 534 might introduce more than a single packet's worth of latency, and since very 535 large packets in streams with a very large number of channels might not fit 536 on a single page. 537 </t> 538 </section> 539 540 <section anchor="end_trimming" title="End Trimming"> 541 <t> 542 The page with the 'end of stream' flag set MAY have a granule position that 543 indicates the page contains less audio data than would normally be returned by 544 decoding up through the final packet. 545 This is used to end the stream somewhere other than an even frame boundary. 546 The granule position of the most recent audio data page with completed packets 547 is used to make this determination, or '0' is used if there were no previous 548 audio data pages with a completed packet. 549 The difference between these granule positions indicates how many samples to 550 keep after decoding the packets that completed on the final page. 551 The remaining samples are discarded. 552 The number of discarded samples SHOULD be no larger than the number decoded 553 from the last packet. 554 </t> 555 </section> 556 557 <section anchor="start_granpos_restrictions" 558 title="Restrictions on the Initial Granule Position"> 559 <t> 560 The granule position of the first audio data page with a completed packet MAY 561 be larger than the number of samples contained in packets that complete on 562 that page, however it MUST NOT be smaller, unless that page has the 'end of 563 stream' flag set. 564 Allowing a granule position larger than the number of samples allows the 565 beginning of a stream to be cropped or a live stream to be joined without 566 rewriting the granule position of all the remaining pages. 567 This means that the PCM sample position just before the first sample to be 568 played MAY be larger than '0'. 569 Synchronization when multiplexing with other logical streams still uses the PCM 570 sample position relative to '0' to compute sample times. 571 This does not affect the behavior of pre-skip: exactly 'pre-skip' samples 572 SHOULD be skipped from the beginning of the decoded output, even if the 573 initial PCM sample position is greater than zero. 574 </t> 575 576 <t> 577 On the other hand, a granule position that is smaller than the number of 578 decoded samples prevents a demuxer from working backwards to assign each 579 packet or each individual sample a valid granule position, since granule 580 positions are non-negative. 581 An implementation MUST treat any stream as invalid if the granule position 582 is smaller than the number of samples contained in packets that complete on 583 the first audio data page with a completed packet, unless that page has the 584 'end of stream' flag set. 585 It MAY defer this action until it decodes the last packet completed on that 586 page. 587 </t> 588 589 <t> 590 If that page has the 'end of stream' flag set, a demuxer MUST treat any stream 591 as invalid if its granule position is smaller than the 'pre-skip' amount. 592 This would indicate that there are more samples to be skipped from the initial 593 decoded output than exist in the stream. 594 If the granule position is smaller than the number of decoded samples produced 595 by the packets that complete on that page, then a demuxer MUST use an initial 596 granule position of '0', and can work forwards from '0' to timestamp 597 individual packets. 598 If the granule position is larger than the number of decoded samples available, 599 then the demuxer MUST still work backwards as described above, even if the 600 'end of stream' flag is set, to determine the initial granule position, and 601 thus the initial PCM sample position. 602 Both of these will be greater than '0' in this case. 603 </t> 604 </section> 605 606 <section anchor="seeking_and_preroll" title="Seeking and Pre-roll"> 607 <t> 608 Seeking in Ogg files is best performed using a bisection search for a page 609 whose granule position corresponds to a PCM position at or before the seek 610 target. 611 With appropriately weighted bisection, accurate seeking can be performed in 612 just one or two bisections on average, even in multi-gigabyte files. 613 See <xref target="seeking"/> for an example of general implementation guidance. 614 </t> 615 616 <t> 617 When seeking within an Ogg Opus stream, an implementation SHOULD start decoding 618 (and discarding the output) at least 3840 samples (80 ms) prior to 619 the seek target in order to ensure that the output audio is correct by the 620 time it reaches the seek target. 621 This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the 622 beginning of the stream. 623 If the point 80 ms prior to the seek target comes before the initial PCM 624 sample position, an implementation SHOULD start decoding from the beginning of 625 the stream, applying pre-skip as normal, regardless of whether the pre-skip is 626 larger or smaller than 80 ms, and then continue to discard samples 627 to reach the seek target (if any). 628 </t> 629 </section> 630 631 </section> 632 633 <section anchor="headers" title="Header Packets"> 634 <t> 635 An Ogg Opus logical stream contains exactly two mandatory header packets: 636 an identification header and a comment header. 637 </t> 638 639 <section anchor="id_header" title="Identification Header"> 640 641 <figure anchor="id_header_packet" title="ID Header Packet" align="center"> 642 <artwork align="center"><![CDATA[ 643 0 1 2 3 644 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 1 645 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 646 | 'O' | 'p' | 'u' | 's' | 647 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 648 | 'H' | 'e' | 'a' | 'd' | 649 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 650 | Version = 1 | Channel Count | Pre-skip | 651 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 652 | Input Sample Rate (Hz) | 653 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 654 | Output Gain (Q7.8 in dB) | Mapping Family| | 655 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ : 656 | | 657 : Optional Channel Mapping Table... : 658 | | 659 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 660 ]]></artwork> 661 </figure> 662 663 <t> 664 The fields in the identification (ID) header have the following meaning: 665 <list style="numbers"> 666 <t>Magic Signature: 667 <vspace blankLines="1"/> 668 This is an 8-octet (64-bit) field that allows codec identification and is 669 human-readable. 670 It contains, in order, the magic numbers: 671 <list style="empty"> 672 <t>0x4F 'O'</t> 673 <t>0x70 'p'</t> 674 <t>0x75 'u'</t> 675 <t>0x73 's'</t> 676 <t>0x48 'H'</t> 677 <t>0x65 'e'</t> 678 <t>0x61 'a'</t> 679 <t>0x64 'd'</t> 680 </list> 681 Starting with "Op" helps distinguish it from audio data packets, as this is an 682 invalid TOC sequence. 683 <vspace blankLines="1"/> 684 </t> 685 <t>Version (8 bits, unsigned): 686 <vspace blankLines="1"/> 687 The version number MUST always be '1' for this version of the encapsulation 688 specification. 689 Implementations SHOULD treat streams where the upper four bits of the version 690 number match that of a recognized specification as backwards-compatible with 691 that specification. 692 That is, the version number can be split into "major" and "minor" version 693 sub-fields, with changes to the "minor" sub-field (in the lower four bits) 694 signaling compatible changes. 695 For example, an implementation of this specification SHOULD accept any stream 696 with a version number of '15' or less, and SHOULD assume any stream with a 697 version number '16' or greater is incompatible. 698 The initial version '1' was chosen to keep implementations from relying on this 699 octet as a null terminator for the "OpusHead" string. 700 <vspace blankLines="1"/> 701 </t> 702 <t>Output Channel Count 'C' (8 bits, unsigned): 703 <vspace blankLines="1"/> 704 This is the number of output channels. 705 This might be different than the number of encoded channels, which can change 706 on a packet-by-packet basis. 707 This value MUST NOT be zero. 708 The maximum allowable value depends on the channel mapping family, and might be 709 as large as 255. 710 See <xref target="channel_mapping"/> for details. 711 <vspace blankLines="1"/> 712 </t> 713 <t>Pre-skip (16 bits, unsigned, little 714 endian): 715 <vspace blankLines="1"/> 716 This is the number of samples (at 48 kHz) to discard from the decoder 717 output when starting playback, and also the number to subtract from a page's 718 granule position to calculate its PCM sample position. 719 When cropping the beginning of existing Ogg Opus streams, a pre-skip of at 720 least 3,840 samples (80 ms) is RECOMMENDED to ensure complete 721 convergence in the decoder. 722 <vspace blankLines="1"/> 723 </t> 724 <t>Input Sample Rate (32 bits, unsigned, little 725 endian): 726 <vspace blankLines="1"/> 727 This is the sample rate of the original input (before encoding), in Hz. 728 This field is <spanx style="emph">not</spanx> the sample rate to use for 729 playback of the encoded data. 730 <vspace blankLines="1"/> 731 Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and 732 20 kHz. 733 Each packet in the stream can have a different audio bandwidth. 734 Regardless of the audio bandwidth, the reference decoder supports decoding any 735 stream at a sample rate of 8, 12, 16, 24, or 48 kHz. 736 The original sample rate of the audio passed to the encoder is not preserved 737 by the lossy compression. 738 <vspace blankLines="1"/> 739 An Ogg Opus player SHOULD select the playback sample rate according to the 740 following procedure: 741 <list style="numbers"> 742 <t>If the hardware supports 48 kHz playback, decode at 48 kHz.</t> 743 <t>Otherwise, if the hardware's highest available sample rate is a supported 744 rate, decode at this sample rate.</t> 745 <t>Otherwise, if the hardware's highest available sample rate is less than 746 48 kHz, decode at the next higher Opus supported rate above the highest 747 available hardware rate and resample.</t> 748 <t>Otherwise, decode at 48 kHz and resample.</t> 749 </list> 750 However, the 'Input Sample Rate' field allows the muxer to pass the sample 751 rate of the original input stream as metadata. 752 This is useful when the user requires the output sample rate to match the 753 input sample rate. 754 For example, when not playing the output, an implementation writing PCM format 755 samples to disk might choose to resample the audio back to the original input 756 sample rate to reduce surprise to the user, who might reasonably expect to get 757 back a file with the same sample rate. 758 <vspace blankLines="1"/> 759 A value of zero indicates 'unspecified'. 760 Muxers SHOULD write the actual input sample rate or zero, but implementations 761 which do something with this field SHOULD take care to behave sanely if given 762 crazy values (e.g., do not actually upsample the output to 10 MHz if 763 requested). 764 Implementations SHOULD support input sample rates between 8 kHz and 765 192 kHz (inclusive). 766 Rates outside this range MAY be ignored by falling back to the default rate of 767 48 kHz instead. 768 <vspace blankLines="1"/> 769 </t> 770 <t>Output Gain (16 bits, signed, little endian): 771 <vspace blankLines="1"/> 772 This is a gain to be applied when decoding. 773 It is 20*log10 of the factor by which to scale the decoder output to achieve 774 the desired playback volume, stored in a 16-bit, signed, two's complement 775 fixed-point value with 8 fractional bits (i.e., 776 Q7.8 <xref target="q-notation"/>). 777 <vspace blankLines="1"/> 778 To apply the gain, an implementation could use 779 <figure align="center"> 780 <artwork align="center"><![CDATA[ 781 sample *= pow(10, output_gain/(20.0*256)) , 782 ]]></artwork> 783 </figure> 784 where output_gain is the raw 16-bit value from the header. 785 <vspace blankLines="1"/> 786 Players and media frameworks SHOULD apply it by default. 787 If a player chooses to apply any volume adjustment or gain modification, such 788 as the R128_TRACK_GAIN (see <xref target="comment_header"/>), the adjustment 789 MUST be applied in addition to this output gain in order to achieve playback 790 at the normalized volume. 791 <vspace blankLines="1"/> 792 A muxer SHOULD set this field to zero, and instead apply any gain prior to 793 encoding, when this is possible and does not conflict with the user's wishes. 794 A nonzero output gain indicates the gain was adjusted after encoding, or that 795 a user wished to adjust the gain for playback while preserving the ability 796 to recover the original signal amplitude. 797 <vspace blankLines="1"/> 798 Although the output gain has enormous range (+/- 128 dB, enough to amplify 799 inaudible sounds to the threshold of physical pain), most applications can 800 only reasonably use a small portion of this range around zero. 801 The large range serves in part to ensure that gain can always be losslessly 802 transferred between OpusHead and R128 gain tags (see below) without 803 saturating. 804 <vspace blankLines="1"/> 805 </t> 806 <t>Channel Mapping Family (8 bits, unsigned): 807 <vspace blankLines="1"/> 808 This octet indicates the order and semantic meaning of the output channels. 809 <vspace blankLines="1"/> 810 Each currently specified value of this octet indicates a mapping family, which 811 defines a set of allowed channel counts, and the ordered set of channel names 812 for each allowed channel count. 813 The details are described in <xref target="channel_mapping"/>. 814 </t> 815 <t>Channel Mapping Table: 816 This table defines the mapping from encoded streams to output channels. 817 Its contents are specified in <xref target="channel_mapping"/>. 818 </t> 819 </list> 820 </t> 821 822 <t> 823 All fields in the ID headers are REQUIRED, except for the channel mapping 824 table, which MUST be omitted when the channel mapping family is 0, but 825 is REQUIRED otherwise. 826 Implementations SHOULD treat a stream as invalid if it contains an ID header 827 that does not have enough data for these fields, even if it contain a valid 828 Magic Signature. 829 Future versions of this specification, even backwards-compatible versions, 830 might include additional fields in the ID header. 831 If an ID header has a compatible major version, but a larger minor version, 832 an implementation MUST NOT treat it as invalid for containing additional data 833 not specified here, provided it still completes on the first page. 834 </t> 835 836 <section anchor="channel_mapping" title="Channel Mapping"> 837 <t> 838 An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly 839 larger number of decoded channels (M + N) to yet another number of 840 output channels (C), which might be larger or smaller than the number of 841 decoded channels. 842 The order and meaning of these channels are defined by a channel mapping, 843 which consists of the 'channel mapping family' octet and, for channel mapping 844 families other than family 0, a channel mapping table, as illustrated in 845 <xref target="channel_mapping_table"/>. 846 </t> 847 848 <figure anchor="channel_mapping_table" title="Channel Mapping Table" 849 align="center"> 850 <artwork align="center"><![CDATA[ 851 0 1 2 3 852 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 1 853 +-+-+-+-+-+-+-+-+ 854 | Stream Count | 855 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 856 | Coupled Count | Channel Mapping... : 857 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 858 ]]></artwork> 859 </figure> 860 861 <t> 862 The fields in the channel mapping table have the following meaning: 863 <list style="numbers" counter="8"> 864 <t>Stream Count 'N' (8 bits, unsigned): 865 <vspace blankLines="1"/> 866 This is the total number of streams encoded in each Ogg packet. 867 This value is necessary to correctly parse the packed Opus packets inside an 868 Ogg packet, as described in <xref target="packet_organization"/>. 869 This value MUST NOT be zero, as without at least one Opus packet with a valid 870 TOC sequence, a demuxer cannot recover the duration of an Ogg packet. 871 <vspace blankLines="1"/> 872 For channel mapping family 0, this value defaults to 1, and is not coded. 873 <vspace blankLines="1"/> 874 </t> 875 <t>Coupled Stream Count 'M' (8 bits, unsigned): 876 This is the number of streams whose decoders are to be configured to produce 877 two channels (stereo). 878 This MUST be no larger than the total number of streams, N. 879 <vspace blankLines="1"/> 880 Each packet in an Opus stream has an internal channel count of 1 or 2, which 881 can change from packet to packet. 882 This is selected by the encoder depending on the bitrate and the audio being 883 encoded. 884 The original channel count of the audio passed to the encoder is not 885 necessarily preserved by the lossy compression. 886 <vspace blankLines="1"/> 887 Regardless of the internal channel count, any Opus stream can be decoded as 888 mono (a single channel) or stereo (two channels) by appropriate initialization 889 of the decoder. 890 The 'coupled stream count' field indicates that the decoders for the first M 891 Opus streams are to be initialized for stereo (two-channel) output, and the 892 remaining (N - M) decoders are to be initialized for mono (a single 893 channel) only. 894 The total number of decoded channels, (M + N), MUST be no larger than 895 255, as there is no way to index more channels than that in the channel 896 mapping. 897 <vspace blankLines="1"/> 898 For channel mapping family 0, this value defaults to (C - 1) 899 (i.e., 0 for mono and 1 for stereo), and is not coded. 900 <vspace blankLines="1"/> 901 </t> 902 <t>Channel Mapping (8*C bits): 903 This contains one octet per output channel, indicating which decoded channel 904 is to be used for each one. 905 Let 'index' be the value of this octet for a particular output channel. 906 This value MUST either be smaller than (M + N), or be the special 907 value 255. 908 If 'index' is less than 2*M, the output MUST be taken from decoding stream 909 ('index'/2) as stereo and selecting the left channel if 'index' is even, and 910 the right channel if 'index' is odd. 911 If 'index' is 2*M or larger, but less than 255, the output MUST be taken from 912 decoding stream ('index' - M) as mono. 913 If 'index' is 255, the corresponding output channel MUST contain pure silence. 914 <vspace blankLines="1"/> 915 The number of output channels, C, is not constrained to match the number of 916 decoded channels (M + N). 917 A single index value MAY appear multiple times, i.e., the same decoded channel 918 might be mapped to multiple output channels. 919 Some decoded channels might not be assigned to any output channel, as well. 920 <vspace blankLines="1"/> 921 For channel mapping family 0, the first index defaults to 0, and if 922 C == 2, the second index defaults to 1. 923 Neither index is coded. 924 </t> 925 </list> 926 </t> 927 928 <t> 929 After producing the output channels, the channel mapping family determines the 930 semantic meaning of each one. 931 There are three defined mapping families in this specification. 932 </t> 933 934 <section anchor="channel_mapping_0" title="Channel Mapping Family 0"> 935 <t> 936 Allowed numbers of channels: 1 or 2. 937 RTP mapping. 938 This is the same channel interpretation as <xref target="RFC7587"/>. 939 </t> 940 <t> 941 <list style="symbols"> 942 <t>1 channel: monophonic (mono).</t> 943 <t>2 channels: stereo (left, right).</t> 944 </list> 945 Special mapping: This channel mapping value also 946 indicates that the contents consists of a single Opus stream that is stereo if 947 and only if C == 2, with stream index 0 mapped to output 948 channel 0 (mono, or left channel) and stream index 1 mapped to 949 output channel 1 (right channel) if stereo. 950 When the 'channel mapping family' octet has this value, the channel mapping 951 table MUST be omitted from the ID header packet. 952 </t> 953 </section> 954 955 <section anchor="channel_mapping_1" title="Channel Mapping Family 1"> 956 <t> 957 Allowed numbers of channels: 1...8. 958 Vorbis channel order (see below). 959 </t> 960 <t> 961 Each channel is assigned to a speaker location in a conventional surround 962 arrangement. 963 Specific locations depend on the number of channels, and are given below 964 in order of the corresponding channel indices. 965 <list style="symbols"> 966 <t>1 channel: monophonic (mono).</t> 967 <t>2 channels: stereo (left, right).</t> 968 <t>3 channels: linear surround (left, center, right)</t> 969 <t>4 channels: quadraphonic (front left, front right, rear left, rear right).</t> 970 <t>5 channels: 5.0 surround (front left, front center, front right, rear left, rear right).</t> 971 <t>6 channels: 5.1 surround (front left, front center, front right, rear left, rear right, LFE).</t> 972 <t>7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE).</t> 973 <t>8 channels: 7.1 surround (front left, front center, front right, side left, side right, rear left, rear right, LFE)</t> 974 </list> 975 </t> 976 <t> 977 This set of surround options and speaker location orderings is the same 978 as those used by the Vorbis codec <xref target="vorbis-mapping"/>. 979 The ordering is different from the one used by the 980 WAVE <xref target="wave-multichannel"/> and 981 Free Lossless Audio Codec (FLAC) <xref target="flac"/> formats, 982 so correct ordering requires permutation of the output channels when decoding 983 to or encoding from those formats. 984 'LFE' here refers to a Low Frequency Effects channel, often mapped to a 985 subwoofer with no particular spatial position. 986 Implementations SHOULD identify 'side' or 'rear' speaker locations with 987 'surround' and 'back' as appropriate when interfacing with audio formats 988 or systems which prefer that terminology. 989 </t> 990 </section> 991 992 <section anchor="channel_mapping_255" 993 title="Channel Mapping Family 255"> 994 <t> 995 Allowed numbers of channels: 1...255. 996 No defined channel meaning. 997 </t> 998 <t> 999 Channels are unidentified. 1000 General-purpose players SHOULD NOT attempt to play these streams. 1001 Offline implementations MAY deinterleave the output into separate PCM files, 1002 one per channel. 1003 Implementations SHOULD NOT produce output for channels mapped to stream index 1004 255 (pure silence) unless they have no other way to indicate the index of 1005 non-silent channels. 1006 </t> 1007 </section> 1008 1009 <section anchor="channel_mapping_undefined" 1010 title="Undefined Channel Mappings"> 1011 <t> 1012 The remaining channel mapping families (2...254) are reserved. 1013 A demuxer implementation encountering a reserved channel mapping family value 1014 SHOULD act as though the value is 255. 1015 </t> 1016 </section> 1017 1018 <section anchor="downmix" title="Downmixing"> 1019 <t> 1020 An Ogg Opus player MUST support any valid channel mapping with a channel 1021 mapping family of 0 or 1, even if the number of channels does not match the 1022 physically connected audio hardware. 1023 Players SHOULD perform channel mixing to increase or reduce the number of 1024 channels as needed. 1025 </t> 1026 1027 <t> 1028 Implementations MAY use the matrices in 1029 Figures <xref target="downmix-matrix-3" format="counter"/> 1030 through <xref target="downmix-matrix-8" format="counter"/> to implement 1031 downmixing from multichannel files using 1032 <xref target="channel_mapping_1">Channel Mapping Family 1</xref>, which are 1033 known to give acceptable results for stereo. 1034 Matrices for 3 and 4 channels are normalized so each coefficient row sums 1035 to 1 to avoid clipping. 1036 For 5 or more channels they are normalized to 2 as a compromise between 1037 clipping and dynamic range reduction. 1038 </t> 1039 <t> 1040 In these matrices the front left and front right channels are generally 1041 passed through directly. 1042 When a surround channel is split between both the left and right stereo 1043 channels, coefficients are chosen so their squares sum to 1, which 1044 helps preserve the perceived intensity. 1045 Rear channels are mixed more diffusely or attenuated to maintain focus 1046 on the front channels. 1047 </t> 1048 1049 <figure anchor="downmix-matrix-3" 1050 title="Stereo downmix matrix for the linear surround channel mapping" 1051 align="center"> 1052 <artwork align="center"><![CDATA[ 1053 L output = ( 0.585786 * left + 0.414214 * center ) 1054 R output = ( 0.414214 * center + 0.585786 * right ) 1055 ]]></artwork> 1056 <postamble> 1057 Exact coefficient values are 1 and 1/sqrt(2), multiplied by 1058 1/(1 + 1/sqrt(2)) for normalization. 1059 </postamble> 1060 </figure> 1061 1062 <figure anchor="downmix-matrix-4" 1063 title="Stereo downmix matrix for the quadraphonic channel mapping" 1064 align="center"> 1065 <artwork align="center"><![CDATA[ 1066 / \ / \ / FL \ 1067 | L output | | 0.422650 0.000000 0.366025 0.211325 | | FR | 1068 | R output | = | 0.000000 0.422650 0.211325 0.366025 | | RL | 1069 \ / \ / \ RR / 1070 ]]></artwork> 1071 <postamble> 1072 Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by 1073 1/(1 + sqrt(3)/2 + 1/2) for normalization. 1074 </postamble> 1075 </figure> 1076 1077 <figure anchor="downmix-matrix-5" 1078 title="Stereo downmix matrix for the 5.0 surround mapping" 1079 align="center"> 1080 <artwork align="center"><![CDATA[ 1081 / FL \ 1082 / \ / \ | FC | 1083 | L | | 0.650802 0.460186 0.000000 0.563611 0.325401 | | FR | 1084 | R | = | 0.000000 0.460186 0.650802 0.325401 0.563611 | | RL | 1085 \ / \ / | RR | 1086 \ / 1087 ]]></artwork> 1088 <postamble> 1089 Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 1090 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2) 1091 for normalization. 1092 </postamble> 1093 </figure> 1094 1095 <figure anchor="downmix-matrix-6" 1096 title="Stereo downmix matrix for the 5.1 surround mapping" 1097 align="center"> 1098 <artwork align="center"><![CDATA[ 1099 /FL \ 1100 / \ / \ |FC | 1101 |L| | 0.529067 0.374107 0.000000 0.458186 0.264534 0.374107 | |FR | 1102 |R| = | 0.000000 0.374107 0.529067 0.264534 0.458186 0.374107 | |RL | 1103 \ / \ / |RR | 1104 \LFE/ 1105 ]]></artwork> 1106 <postamble> 1107 Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 1108 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1/sqrt(2)) 1109 for normalization. 1110 </postamble> 1111 </figure> 1112 1113 <figure anchor="downmix-matrix-7" 1114 title="Stereo downmix matrix for the 6.1 surround mapping" 1115 align="center"> 1116 <artwork align="center"><![CDATA[ 1117 / \ 1118 | 0.455310 0.321953 0.000000 0.394310 0.227655 0.278819 0.321953 | 1119 | 0.000000 0.321953 0.455310 0.227655 0.394310 0.278819 0.321953 | 1120 \ / 1121 ]]></artwork> 1122 <postamble> 1123 Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and 1124 sqrt(3)/2/sqrt(2), multiplied by 1125 2/(1 + 1/sqrt(2) + sqrt(3)/2 + 1/2 + 1126 sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization. 1127 The coefficients are in the same order as in <xref target="channel_mapping_1" />, 1128 and the matrices above. 1129 </postamble> 1130 </figure> 1131 1132 <figure anchor="downmix-matrix-8" 1133 title="Stereo downmix matrix for the 7.1 surround mapping" 1134 align="center"> 1135 <artwork align="center"><![CDATA[ 1136 / \ 1137 | .388631 .274804 .000000 .336565 .194316 .336565 .194316 .274804 | 1138 | .000000 .274804 .388631 .194316 .336565 .194316 .336565 .274804 | 1139 \ / 1140 ]]></artwork> 1141 <postamble> 1142 Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by 1143 2/(2 + 2/sqrt(2) + sqrt(3)) for normalization. 1144 The coefficients are in the same order as in <xref target="channel_mapping_1" />, 1145 and the matrices above. 1146 </postamble> 1147 </figure> 1148 1149 </section> 1150 1151 </section> <!-- end channel_mapping_table --> 1152 1153 </section> <!-- end id_header --> 1154 1155 <section anchor="comment_header" title="Comment Header"> 1156 1157 <figure anchor="comment_header_packet" title="Comment Header Packet" 1158 align="center"> 1159 <artwork align="center"><![CDATA[ 1160 0 1 2 3 1161 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 1162 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1163 | 'O' | 'p' | 'u' | 's' | 1164 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1165 | 'T' | 'a' | 'g' | 's' | 1166 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1167 | Vendor String Length | 1168 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1169 | | 1170 : Vendor String... : 1171 | | 1172 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1173 | User Comment List Length | 1174 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1175 | User Comment #0 String Length | 1176 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1177 | | 1178 : User Comment #0 String... : 1179 | | 1180 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1181 | User Comment #1 String Length | 1182 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 1183 : : 1184 ]]></artwork> 1185 </figure> 1186 1187 <t> 1188 The comment header consists of a 64-bit magic signature, followed by data in 1189 the same format as the <xref target="vorbis-comment"/> header used in Ogg 1190 Vorbis, except (like Ogg Theora and Speex) the final "framing bit" specified 1191 in the Vorbis spec is not present. 1192 <list style="numbers"> 1193 <t>Magic Signature: 1194 <vspace blankLines="1"/> 1195 This is an 8-octet (64-bit) field that allows codec identification and is 1196 human-readable. 1197 It contains, in order, the magic numbers: 1198 <list style="empty"> 1199 <t>0x4F 'O'</t> 1200 <t>0x70 'p'</t> 1201 <t>0x75 'u'</t> 1202 <t>0x73 's'</t> 1203 <t>0x54 'T'</t> 1204 <t>0x61 'a'</t> 1205 <t>0x67 'g'</t> 1206 <t>0x73 's'</t> 1207 </list> 1208 Starting with "Op" helps distinguish it from audio data packets, as this is an 1209 invalid TOC sequence. 1210 <vspace blankLines="1"/> 1211 </t> 1212 <t>Vendor String Length (32 bits, unsigned, little endian): 1213 <vspace blankLines="1"/> 1214 This field gives the length of the following vendor string, in octets. 1215 It MUST NOT indicate that the vendor string is longer than the rest of the 1216 packet. 1217 <vspace blankLines="1"/> 1218 </t> 1219 <t>Vendor String (variable length, UTF-8 vector): 1220 <vspace blankLines="1"/> 1221 This is a simple human-readable tag for vendor information, encoded as a UTF-8 1222 string <xref target="RFC3629"/>. 1223 No terminating null octet is necessary. 1224 <vspace blankLines="1"/> 1225 This tag is intended to identify the codec encoder and encapsulation 1226 implementations, for tracing differences in technical behavior. 1227 User-facing applications can use the 'ENCODER' user comment tag to identify 1228 themselves. 1229 <vspace blankLines="1"/> 1230 </t> 1231 <t>User Comment List Length (32 bits, unsigned, little endian): 1232 <vspace blankLines="1"/> 1233 This field indicates the number of user-supplied comments. 1234 It MAY indicate there are zero user-supplied comments, in which case there are 1235 no additional fields in the packet. 1236 It MUST NOT indicate that there are so many comments that the comment string 1237 lengths would require more data than is available in the rest of the packet. 1238 <vspace blankLines="1"/> 1239 </t> 1240 <t>User Comment #i String Length (32 bits, unsigned, little endian): 1241 <vspace blankLines="1"/> 1242 This field gives the length of the following user comment string, in octets. 1243 There is one for each user comment indicated by the 'user comment list length' 1244 field. 1245 It MUST NOT indicate that the string is longer than the rest of the packet. 1246 <vspace blankLines="1"/> 1247 </t> 1248 <t>User Comment #i String (variable length, UTF-8 vector): 1249 <vspace blankLines="1"/> 1250 This field contains a single user comment encoded as a UTF-8 1251 string <xref target="RFC3629"/>. 1252 There is one for each user comment indicated by the 'user comment list length' 1253 field. 1254 </t> 1255 </list> 1256 </t> 1257 1258 <t> 1259 The vendor string length and user comment list length are REQUIRED, and 1260 implementations SHOULD treat a stream as invalid if it contains a comment 1261 header that does not have enough data for these fields, or that does not 1262 contain enough data for the corresponding vendor string or user comments they 1263 describe. 1264 Making this check before allocating the associated memory to contain the data 1265 helps prevent a possible Denial-of-Service (DoS) attack from small comment 1266 headers that claim to contain strings longer than the entire packet or more 1267 user comments than than could possibly fit in the packet. 1268 </t> 1269 1270 <t> 1271 Immediately following the user comment list, the comment header MAY 1272 contain zero-padding or other binary data which is not specified here. 1273 If the least-significant bit of the first byte of this data is 1, then editors 1274 SHOULD preserve the contents of this data when updating the tags, but if this 1275 bit is 0, all such data MAY be treated as padding, and truncated or discarded 1276 as desired. 1277 This allows informal experimentation with the format of this binary data until 1278 it can be specified later. 1279 </t> 1280 1281 <t> 1282 The comment header can be arbitrarily large and might be spread over a large 1283 number of Ogg pages. 1284 Implementations MUST avoid attempting to allocate excessive amounts of memory 1285 when presented with a very large comment header. 1286 To accomplish this, implementations MAY treat a stream as invalid if it has a 1287 comment header larger than 125,829,120 octets (120 MB), and MAY 1288 ignore individual comments that are not fully contained within the first 1289 61,440 octets of the comment header. 1290 </t> 1291 1292 <section anchor="comment_format" title="Tag Definitions"> 1293 <t> 1294 The user comment strings follow the NAME=value format described by 1295 <xref target="vorbis-comment"/> with the same recommended tag names: 1296 ARTIST, TITLE, DATE, ALBUM, and so on. 1297 </t> 1298 <t> 1299 Two new comment tags are introduced here: 1300 </t> 1301 1302 <t>First, an optional gain for track normalization:</t> 1303 <figure align="center"> 1304 <artwork align="left"><![CDATA[ 1305 R128_TRACK_GAIN=-573 1306 ]]></artwork> 1307 </figure> 1308 <t> 1309 representing the volume shift needed to normalize the track's volume 1310 during isolated playback, in random shuffle, and so on. 1311 The gain is a Q7.8 fixed point number in dB, as in the ID header's 'output 1312 gain' field. 1313 This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in 1314 Vorbis <xref target="replay-gain"/>, except that the normal volume 1315 reference is the <xref target="EBU-R128"/> standard. 1316 </t> 1317 <t>Second, an optional gain for album normalization:</t> 1318 <figure align="center"> 1319 <artwork align="left"><![CDATA[ 1320 R128_ALBUM_GAIN=111 1321 ]]></artwork> 1322 </figure> 1323 <t> 1324 representing the volume shift needed to normalize the overall volume when 1325 played as part of a particular collection of tracks. 1326 The gain is also a Q7.8 fixed point number in dB, as in the ID header's 1327 'output gain' field. 1328 The values '-573' and '111' given here are just examples. 1329 </t> 1330 <t> 1331 An Ogg Opus stream MUST NOT have more than one of each of these tags, and if 1332 present their values MUST be an integer from -32768 to 32767, inclusive, 1333 represented in ASCII as a base 10 number with no whitespace. 1334 A leading '+' or '-' character is valid. 1335 Leading zeros are also permitted, but the value MUST be represented by 1336 no more than 6 characters. 1337 Other non-digit characters MUST NOT be present. 1338 </t> 1339 <t> 1340 If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent 1341 the R128 normalization gain relative to the 'output gain' field specified 1342 in the ID header. 1343 If a player chooses to make use of the R128_TRACK_GAIN tag or the 1344 R128_ALBUM_GAIN tag, it MUST apply those gains 1345 <spanx style="emph">in addition</spanx> to the 'output gain' value. 1346 If a tool modifies the ID header's 'output gain' field, it MUST also update or 1347 remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present. 1348 A muxer SHOULD place the gain it wants other tools to use by default into the 1349 'output gain' field, and not the comment tag. 1350 </t> 1351 <t> 1352 To avoid confusion with multiple normalization schemes, an Opus comment header 1353 SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK, 1354 REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags, unless they are only 1355 to be used in some context where there is guaranteed to be no such confusion. 1356 <xref target="EBU-R128"/> normalization is preferred to the earlier 1357 REPLAYGAIN schemes because of its clear definition and adoption by industry. 1358 Peak normalizations are difficult to calculate reliably for lossy codecs 1359 because of variation in excursion heights due to decoder differences. 1360 In the authors' investigations they were not applied consistently or broadly 1361 enough to merit inclusion here. 1362 </t> 1363 </section> <!-- end comment_format --> 1364 </section> <!-- end comment_header --> 1365 1366 </section> <!-- end headers --> 1367 1368 <section anchor="packet_size_limits" title="Packet Size Limits"> 1369 <t> 1370 Technically, valid Opus packets can be arbitrarily large due to the padding 1371 format, although the amount of non-padding data they can contain is bounded. 1372 These packets might be spread over a similarly enormous number of Ogg pages. 1373 When encoding, implementations SHOULD limit the use of padding in audio data 1374 packets to no more than is necessary to make a variable bitrate (VBR) stream 1375 constant bitrate (CBR), unless they have no reasonable way to determine what 1376 is necessary. 1377 Demuxers SHOULD treat audio data packets as invalid (treat them as if they were 1378 malformed Opus packets with an invalid TOC sequence) if they are larger than 1379 61,440 octets per Opus stream, unless they have a specific reason for 1380 allowing extra padding. 1381 Such packets necessarily contain more padding than needed to make a stream CBR. 1382 Demuxers MUST avoid attempting to allocate excessive amounts of memory when 1383 presented with a very large packet. 1384 Demuxers MAY treat audio data packets as invalid or partially process them if 1385 they are larger than 61,440 octets in an Ogg Opus stream with channel 1386 mapping families 0 or 1. 1387 Demuxers MAY treat audio data packets as invalid or partially process them in 1388 any Ogg Opus stream if the packet is larger than 61,440 octets and also 1389 larger than 7,680 octets per Opus stream. 1390 The presence of an extremely large packet in the stream could indicate a 1391 memory exhaustion attack or stream corruption. 1392 </t> 1393 <t> 1394 In an Ogg Opus stream, the largest possible valid packet that does not use 1395 padding has a size of (61,298*N - 2) octets. 1396 With 255 streams, this is 15,630,988 octets and can 1397 span up to 61,298 Ogg pages, all but one of which will have a granule 1398 position of -1. 1399 This is of course a very extreme packet, consisting of 255 streams, each 1400 containing 120 ms of audio encoded as 2.5 ms frames, each frame 1401 using the maximum possible number of octets (1275) and stored in the least 1402 efficient manner allowed (a VBR code 3 Opus packet). 1403 Even in such a packet, most of the data will be zeros as 2.5 ms frames 1404 cannot actually use all 1275 octets. 1405 </t> 1406 <t> 1407 The largest packet consisting of entirely useful data is 1408 (15,326*N - 2) octets. 1409 This corresponds to 120 ms of audio encoded as 10 ms frames in either 1410 SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little 1411 sense for the quality achieved. 1412 </t> 1413 <t> 1414 A more reasonable limit is (7,664*N - 2) octets. 1415 This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode 1416 frames, with a total bitrate just under 511 kbps (not counting the Ogg 1417 encapsulation overhead). 1418 For channel mapping family 1, N=8 provides a reasonable upper bound, as it 1419 allows for each of the 8 possible output channels to be decoded from a 1420 separate stereo Opus stream. 1421 This gives a size of 61,310 octets, which is rounded up to a multiple of 1422 1,024 octets to yield the audio data packet size of 61,440 octets 1423 that any implementation is expected to be able to process successfully. 1424 </t> 1425 </section> 1426 1427 <section anchor="encoder" title="Encoder Guidelines"> 1428 <t> 1429 When encoding Opus streams, Ogg muxers SHOULD take into account the 1430 algorithmic delay of the Opus encoder. 1431 </t> 1432 <t> 1433 In encoders derived from the reference 1434 implementation <xref target="RFC6716"/>, the number of samples can be 1435 queried with: 1436 </t> 1437 <figure align="center"> 1438 <artwork align="center"><![CDATA[ 1439 opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples)); 1440 ]]></artwork> 1441 </figure> 1442 <t> 1443 To achieve good quality in the very first samples of a stream, implementations 1444 MAY use linear predictive coding (LPC) extrapolation to generate at least 120 1445 extra samples at the beginning to avoid the Opus encoder having to encode a 1446 discontinuous signal. 1447 For more information on linear prediction, see 1448 <xref target="linear-prediction"/>. 1449 For an input file containing 'length' samples, the implementation SHOULD set 1450 the pre-skip header value to (delay_samples + extra_samples), encode 1451 at least (length + delay_samples + extra_samples) 1452 samples, and set the granule position of the last page to 1453 (length + delay_samples + extra_samples). 1454 This ensures that the encoded file has the same duration as the original, with 1455 no time offset. The best way to pad the end of the stream is to also use LPC 1456 extrapolation, but zero-padding is also acceptable. 1457 </t> 1458 1459 <section anchor="lpc" title="LPC Extrapolation"> 1460 <t> 1461 The first step in LPC extrapolation is to compute linear prediction 1462 coefficients. <xref target="lpc-sample"/> 1463 When extending the end of the signal, order-N (typically with N ranging from 8 1464 to 40) LPC analysis is performed on a window near the end of the signal. 1465 The last N samples are used as memory to an infinite impulse response (IIR) 1466 filter. 1467 </t> 1468 <t> 1469 The filter is then applied on a zero input to extrapolate the end of the signal. 1470 Let a(k) be the kth LPC coefficient and x(n) be the nth sample of the signal, 1471 each new sample past the end of the signal is computed as: 1472 </t> 1473 <figure align="center"> 1474 <artwork align="center"><![CDATA[ 1475 N 1476 --- 1477 x(n) = \ a(k)*x(n-k) 1478 / 1479 --- 1480 k=1 1481 ]]></artwork> 1482 </figure> 1483 <t> 1484 The process is repeated independently for each channel. 1485 It is possible to extend the beginning of the signal by applying the same 1486 process backward in time. 1487 When extending the beginning of the signal, it is best to apply a "fade in" to 1488 the extrapolated signal, e.g. by multiplying it by a half-Hanning window 1489 <xref target="hanning"/>. 1490 </t> 1491 1492 </section> 1493 1494 <section anchor="continuous_chaining" title="Continuous Chaining"> 1495 <t> 1496 In some applications, such as Internet radio, it is desirable to cut a long 1497 stream into smaller chains, e.g. so the comment header can be updated. 1498 This can be done simply by separating the input streams into segments and 1499 encoding each segment independently. 1500 The drawback of this approach is that it creates a small discontinuity 1501 at the boundary due to the lossy nature of Opus. 1502 A muxer MAY avoid this discontinuity by using the following procedure: 1503 <list style="numbers"> 1504 <t>Encode the last frame of the first segment as an independent frame by 1505 turning off all forms of inter-frame prediction. 1506 De-emphasis is allowed.</t> 1507 <t>Set the granule position of the last page to a point near the end of the 1508 last frame.</t> 1509 <t>Begin the second segment with a copy of the last frame of the first 1510 segment.</t> 1511 <t>Set the pre-skip value of the second stream in such a way as to properly 1512 join the two streams.</t> 1513 <t>Continue the encoding process normally from there, without any reset to 1514 the encoder.</t> 1515 </list> 1516 </t> 1517 <t> 1518 In encoders derived from the reference implementation, inter-frame prediction 1519 can be turned off by calling: 1520 </t> 1521 <figure align="center"> 1522 <artwork align="center"><![CDATA[ 1523 opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1)); 1524 ]]></artwork> 1525 </figure> 1526 <t> 1527 For best results, this implementation requires that prediction be explicitly 1528 enabled again before resuming normal encoding, even after a reset. 1529 </t> 1530 1531 </section> 1532 1533 </section> 1534 1535 <section anchor="implementation" title="Implementation Status"> 1536 <t> 1537 A brief summary of major implementations of this draft is available 1538 at <eref target="https://wiki.xiph.org/OggOpusImplementation"/>, 1539 along with their status. 1540 </t> 1541 <t> 1542 [Note to RFC Editor: please remove this entire section before 1543 final publication per <xref target="RFC6982"/>, along with 1544 its references.] 1545 </t> 1546 </section> 1547 1548 <section anchor="security" title="Security Considerations"> 1549 <t> 1550 Implementations of the Opus codec need to take appropriate security 1551 considerations into account, as outlined in <xref target="RFC4732"/>. 1552 This is just as much a problem for the container as it is for the codec itself. 1553 Malicious payloads and/or input streams can be used to attack codec 1554 implementations. 1555 Implementations MUST NOT overrun their allocated memory nor consume excessive 1556 resources when decoding payloads or processing input streams. 1557 Although problems in encoding applications are typically rarer, this still 1558 applies to a muxer, as vulnerabilities would allow an attacker to attack 1559 transcoding gateways. 1560 </t> 1561 1562 <t> 1563 Header parsing code contains the most likely area for potential overruns. 1564 It is important for implementations to ensure their buffers contain enough 1565 data for all of the required fields before attempting to read it (for example, 1566 for all of the channel map data in the ID header). 1567 Implementations would do well to validate the indices of the channel map, also, 1568 to ensure they meet all of the restrictions outlined in 1569 <xref target="channel_mapping"/>, in order to avoid attempting to read data 1570 from channels that do not exist. 1571 </t> 1572 1573 <t> 1574 To avoid excessive resource usage, we advise implementations to be especially 1575 wary of streams that might cause them to process far more data than was 1576 actually transmitted. 1577 For example, a relatively small comment header may contain values for the 1578 string lengths or user comment list length that imply that it is many 1579 gigabytes in size. 1580 Even computing the size of the required buffer could overflow a 32-bit integer, 1581 and actually attempting to allocate such a buffer before verifying it would be 1582 a reasonable size is a bad idea. 1583 After reading the user comment list length, implementations might wish to 1584 verify that the header contains at least the minimum amount of data for that 1585 many comments (4 additional octets per comment, to indicate each has a 1586 length of zero) before proceeding any further, again taking care to avoid 1587 overflow in these calculations. 1588 If allocating an array of pointers to point at these strings, the size of the 1589 pointers may be larger than 4 octets, potentially requiring a separate 1590 overflow check. 1591 </t> 1592 1593 <t> 1594 Another bug in this class we have observed more than once involves the handling 1595 of invalid data at the end of a stream. 1596 Often, implementations will seek to the end of a stream to locate the last 1597 timestamp in order to compute its total duration. 1598 If they do not find a valid capture pattern and Ogg page from the desired 1599 logical stream, they will back up and try again. 1600 If care is not taken to avoid re-scanning data that was already scanned, this 1601 search can quickly devolve into something with a complexity that is quadratic 1602 in the amount of invalid data. 1603 </t> 1604 1605 <t> 1606 In general when seeking, implementations will wish to be cautious about the 1607 effects of invalid granule position values, and ensure all algorithms will 1608 continue to make progress and eventually terminate, even if these are missing 1609 or out-of-order. 1610 </t> 1611 1612 <t> 1613 Like most other container formats, Ogg Opus streams SHOULD NOT be used with 1614 insecure ciphers or cipher modes that are vulnerable to known-plaintext 1615 attacks. 1616 Elements such as the Ogg page capture pattern and the magic signatures in the 1617 ID header and the comment header all have easily predictable values, in 1618 addition to various elements of the codec data itself. 1619 </t> 1620 </section> 1621 1622 <section anchor="content_type" title="Content Type"> 1623 <t> 1624 An "Ogg Opus file" consists of one or more sequentially multiplexed segments, 1625 each containing exactly one Ogg Opus stream. 1626 The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg". 1627 </t> 1628 1629 <t> 1630 If more specificity is desired, one MAY indicate the presence of Opus streams 1631 using the codecs parameter defined in <xref target="RFC6381"/> and 1632 <xref target="RFC5334"/>, e.g., 1633 </t> 1634 <figure> 1635 <artwork align="center"><![CDATA[ 1636 audio/ogg; codecs=opus 1637 ]]></artwork> 1638 </figure> 1639 <t> 1640 for an Ogg Opus file. 1641 </t> 1642 1643 <t> 1644 The RECOMMENDED filename extension for Ogg Opus files is '.opus'. 1645 </t> 1646 1647 <t> 1648 When Opus is concurrently multiplexed with other streams in an Ogg container, 1649 one SHOULD use one of the "audio/ogg", "video/ogg", or "application/ogg" 1650 mime-types, as defined in <xref target="RFC5334"/>. 1651 Such streams are not strictly "Ogg Opus files" as described above, 1652 since they contain more than a single Opus stream per sequentially 1653 multiplexed segment, e.g. video or multiple audio tracks. 1654 In such cases the the '.opus' filename extension is NOT RECOMMENDED. 1655 </t> 1656 1657 <t> 1658 In either case, this document updates <xref target="RFC5334"/> 1659 to add 'opus' as a codecs parameter value with char[8]: 'OpusHead' 1660 as Codec Identifier. 1661 </t> 1662 </section> 1663 1664 <section anchor="iana" title="IANA Considerations"> 1665 <t> 1666 This document updates the IANA Media Types registry to add .opus 1667 as a file extension for "audio/ogg", and to add itself as a reference 1668 alongside <xref target="RFC5334"/> for "audio/ogg", "video/ogg", and 1669 "application/ogg" Media Types. 1670 </t> 1671 <t> 1672 This document defines a new registry "Opus Channel Mapping Families" to 1673 indicate how the semantic meanings of the channels in a multi-channel Opus 1674 stream are described. 1675 IANA is requested to create a new name space of "Opus Channel Mapping 1676 Families". 1677 This will be a new registry on the IANA Matrix, and not a subregistry of an 1678 existing registry. 1679 Modifications to this registry follow the "Specification Required" registration 1680 policy as defined in <xref target="RFC5226"/>. 1681 Each registry entry consists of a Channel Mapping Family Number, which is 1682 specified in decimal in the range 0 to 255, inclusive, and a Reference (or 1683 list of references) 1684 Each Reference must point to sufficient documentation to describe what 1685 information is coded in the Opus identification header for this channel 1686 mapping family, how a demuxer determines the Stream Count ('N') and Coupled 1687 Stream Count ('M') from this information, and how it determines the proper 1688 interpretation of each of the decoded channels. 1689 </t> 1690 <t> 1691 This document defines three initial assignments for this registry. 1692 </t> 1693 <texttable> 1694 <ttcol>Value</ttcol><ttcol>Reference</ttcol> 1695 <c>0</c><c>[RFCXXXX] <xref target="channel_mapping_0"/></c> 1696 <c>1</c><c>[RFCXXXX] <xref target="channel_mapping_1"/></c> 1697 <c>255</c><c>[RFCXXXX] <xref target="channel_mapping_255"/></c> 1698 </texttable> 1699 <t> 1700 The designated expert will determine if the Reference points to a specification 1701 that meets the requirements for permanence and ready availability laid out 1702 in <xref target="RFC5226"/> and that it specifies the information 1703 described above with sufficient clarity to allow interoperable 1704 implementations. 1705 </t> 1706 </section> 1707 1708 <section anchor="Acknowledgments" title="Acknowledgments"> 1709 <t> 1710 Thanks to Ben Campbell, Joel M. Halpern, Mark Harris, Greg Maxwell, 1711 Christopher "Monty" Montgomery, Jean-Marc Valin, Stephan Wenger, and Mo Zanaty 1712 for their valuable contributions to this document. 1713 Additional thanks to Andrew D'Addesio, Greg Maxwell, and Vincent Penquerc'h for 1714 their feedback based on early implementations. 1715 </t> 1716 </section> 1717 1718 <section title="RFC Editor Notes"> 1719 <t> 1720 In <xref target="iana"/>, "RFCXXXX" is to be replaced with the RFC number 1721 assigned to this draft. 1722 </t> 1723 </section> 1724 1725 </middle> 1726 <back> 1727 <references title="Normative References"> 1728 &rfc2119; 1729 &rfc3533; 1730 &rfc3629; 1731 &rfc5226; 1732 &rfc5334; 1733 &rfc6381; 1734 &rfc6716; 1735 1736 <reference anchor="EBU-R128" target="https://tech.ebu.ch/loudness"> 1737 <front> 1738 <title>Loudness Recommendation EBU R128</title> 1739 <author> 1740 <organization>EBU Technical Committee</organization> 1741 </author> 1742 <date month="August" year="2011"/> 1743 </front> 1744 </reference> 1745 1746 <reference anchor="vorbis-comment" 1747 target="https://www.xiph.org/vorbis/doc/v-comment.html"> 1748 <front> 1749 <title>Ogg Vorbis I Format Specification: Comment Field and Header 1750 Specification</title> 1751 <author initials="C." surname="Montgomery" 1752 fullname="Christopher "Monty" Montgomery"/> 1753 <date month="July" year="2002"/> 1754 </front> 1755 </reference> 1756 1757 </references> 1758 1759 <references title="Informative References"> 1760 1761 <!--?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml"?--> 1762 &rfc4732; 1763 &rfc6982; 1764 &rfc7587; 1765 1766 <reference anchor="flac" 1767 target="https://xiph.org/flac/format.html"> 1768 <front> 1769 <title>FLAC - Free Lossless Audio Codec Format Description</title> 1770 <author initials="J." surname="Coalson" fullname="Josh Coalson"/> 1771 <date month="January" year="2008"/> 1772 </front> 1773 </reference> 1774 1775 <reference anchor="hanning" 1776 target="https://en.wikipedia.org/w/index.php?title=Window_function&oldid=703074467#Hann_.28Hanning.29_window"> 1777 <front> 1778 <title>Hann window</title> 1779 <author> 1780 <organization>Wikipedia</organization> 1781 </author> 1782 <date month="February" year="2016"/> 1783 </front> 1784 </reference> 1785 1786 <reference anchor="linear-prediction" 1787 target="https://en.wikipedia.org/w/index.php?title=Linear_predictive_coding&oldid=687498962"> 1788 <front> 1789 <title>Linear Predictive Coding</title> 1790 <author> 1791 <organization>Wikipedia</organization> 1792 </author> 1793 <date month="October" year="2015"/> 1794 </front> 1795 </reference> 1796 1797 <reference anchor="lpc-sample" 1798 target="https://svn.xiph.org/trunk/vorbis/lib/lpc.c"> 1799 <front> 1800 <title>Autocorrelation LPC coeff generation algorithm 1801 (Vorbis source code)</title> 1802 <author initials="J." surname="Degener" fullname="Jutta Degener"/> 1803 <author initials="C." surname="Bormann" fullname="Carsten Bormann"/> 1804 <date month="November" year="1994"/> 1805 </front> 1806 </reference> 1807 1808 <reference anchor="q-notation" 1809 target="https://en.wikipedia.org/w/index.php?title=Q_%28number_format%29&oldid=697252615"> 1810 <front> 1811 <title>Q (number format)</title> 1812 <author><organization>Wikipedia</organization></author> 1813 <date month="December" year="2015"/> 1814 </front> 1815 </reference> 1816 1817 <reference anchor="replay-gain" 1818 target="https://wiki.xiph.org/VorbisComment#Replay_Gain"> 1819 <front> 1820 <title>VorbisComment: Replay Gain</title> 1821 <author initials="C." surname="Parker" fullname="Conrad Parker"/> 1822 <author initials="M." surname="Leese" fullname="Martin Leese"/> 1823 <date month="June" year="2009"/> 1824 </front> 1825 </reference> 1826 1827 <reference anchor="seeking" 1828 target="https://wiki.xiph.org/Seeking"> 1829 <front> 1830 <title>Granulepos Encoding and How Seeking Really Works</title> 1831 <author initials="S." surname="Pfeiffer" fullname="Silvia Pfeiffer"/> 1832 <author initials="C." surname="Parker" fullname="Conrad Parker"/> 1833 <author initials="G." surname="Maxwell" fullname="Greg Maxwell"/> 1834 <date month="May" year="2012"/> 1835 </front> 1836 </reference> 1837 1838 <reference anchor="vorbis-mapping" 1839 target="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9"> 1840 <front> 1841 <title>The Vorbis I Specification, Section 4.3.9 Output Channel Order</title> 1842 <author initials="C." surname="Montgomery" 1843 fullname="Christopher "Monty" Montgomery"/> 1844 <date month="January" year="2010"/> 1845 </front> 1846 </reference> 1847 1848 <reference anchor="vorbis-trim" 1849 target="https://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-132000A.2"> 1850 <front> 1851 <title>The Vorbis I Specification, Appendix A: Embedding Vorbis 1852 into an Ogg stream</title> 1853 <author initials="C." surname="Montgomery" 1854 fullname="Christopher "Monty" Montgomery"/> 1855 <date month="November" year="2008"/> 1856 </front> 1857 </reference> 1858 1859 <reference anchor="wave-multichannel" 1860 target="http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx"> 1861 <front> 1862 <title>Multiple Channel Audio Data and WAVE Files</title> 1863 <author> 1864 <organization>Microsoft Corporation</organization> 1865 </author> 1866 <date month="March" year="2007"/> 1867 </front> 1868 </reference> 1869 1870 </references> 1871 1872 </back> 1873 </rfc> 1874
