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1<?xml version="1.0" encoding="utf-8"?>
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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'>
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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@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@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@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>
102This document defines the Ogg encapsulation for the Opus interactive speech and
103 audio codec.
104This 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>
113The IETF Opus codec is a low-latency audio codec optimized for both voice and
114 general-purpose audio.
115See <xref target="RFC6716"/> for technical details.
116This document defines the encapsulation of Opus in a continuous, logical Ogg
117 bitstream&nbsp;<xref target="RFC3533"/>.
118Ogg 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.
122It 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>
128Ogg bitstreams are made up of a series of 'pages', each of which contains data
129 from one or more 'packets'.
130Pages are the fundamental unit of multiplexing in an Ogg stream.
131Each 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.
135A single page can contain up to 65,025 octets of packet data from up to 255
136 different packets.
137Packets 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).
139Each 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.
142A 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>
146This 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.
149It does not attempt to describe or specify the existing Ogg container format.
150Readers 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>
158The 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>
167An 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>
194There are two mandatory header packets.
195The first packet in the logical Ogg bitstream MUST contain the identification
196 (ID) header, which uniquely identifies a stream as Opus audio.
197The format of this header is defined in <xref target="id_header"/>.
198It is placed alone (without any other packet data) on the first page of
199 the logical Ogg bitstream, and completes on that page.
200This page has its 'beginning of stream' flag set.
201</t>
202<t>
203The second packet in the logical Ogg bitstream MUST contain the comment header,
204 which contains user-supplied metadata.
205The format of this header is defined in <xref target="comment_header"/>.
206It MAY span multiple pages, beginning on the second page of the logical
207 stream.
208However many pages it spans, the comment header packet MUST finish the page on
209 which it completes.
210</t>
211<t>
212All subsequent pages are audio data pages, and the Ogg packets they contain are
213 audio data packets.
214Each 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.
217The 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>
222The first (N&nbsp;-&nbsp;1) Opus packets, if any, are packed one after another
223 into the Ogg packet, using the self-delimiting framing from Appendix&nbsp;B of
224 <xref target="RFC6716"/>.
225The remaining Opus packet is packed at the end of the Ogg packet using the
226 regular, undelimited framing from Section&nbsp;3 of <xref target="RFC6716"/>.
227All of the Opus packets in a single Ogg packet MUST be constrained to have the
228 same duration.
229An 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>
235The 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&nbsp;3.1
238 of&nbsp;<xref target="RFC6716"/>.
239The coding mode is one of SILK, Hybrid, or Constrained Energy Lapped Transform
240 (CELT).
241The combination of coding mode, audio bandwidth, and frame size is referred to
242 as the configuration of an Opus packet.
243</t>
244<t>
245Packets are placed into Ogg pages in order until the end of stream.
246Audio data packets might span page boundaries.
247The 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.
251If 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.
261An implementation MUST treat a zero-octet audio data packet as if it were a
262 malformed Opus packet as described in
263 Section&nbsp;3.4 of&nbsp;<xref target="RFC6716"/>.
264</t>
265<t>
266A 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'.
269There 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.
271However, demuxers might encounter such streams, possibly as the result of a
272 transfer that did not complete or of corruption.
273If 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.
284There 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>
291The granule position MUST be zero for the ID header page and the
292 page where the comment header completes.
293That 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>
297The 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.
300The 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>
305A 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>
311The granule position of an audio data page is in units of PCM audio samples at
312 a fixed rate of 48&nbsp;kHz (per channel; a stereo stream's granule position
313 does not increment at twice the speed of a mono stream).
314It 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&nbsp;kHz.
317Therefore, the value in the granule position field always counts samples
318 assuming a 48&nbsp;kHz decoding rate, and the rest of this specification makes
319 the same assumption.
320</t>
321
322<t>
323The duration of an Opus packet as defined in <xref target="RFC6716"/> can be
324 any multiple of 2.5&nbsp;ms, up to a maximum of 120&nbsp;ms.
325This duration is encoded in the TOC sequence at the beginning of each packet.
326The number of samples returned by a decoder corresponds to this duration
327 exactly, even for the first few packets.
328For example, a 20&nbsp;ms packet fed to a decoder running at 48&nbsp;kHz will
329 always return 960&nbsp;samples.
330A 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.
334The one exception is the last page in the stream, as described below.
335</t>
336
337<t>
338All 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.
342This guarantees that a demuxer can assign individual packets the same granule
343 position when working forwards as when working backwards.
344For 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>
349In 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.
353Implementations 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>
358Only gaps that are a multiple of 2.5&nbsp;ms are repairable, as these are the
359 only durations that can be created by packet loss or discontinuous
360 transmission.
361Muxers need not handle other gap sizes.
362Creating the necessary packets involves synthesizing a TOC byte (defined in
363Section&nbsp;3.1 of&nbsp;<xref target="RFC6716"/>)&mdash;and whatever
364 additional internal framing is needed&mdash;to indicate the packet duration
365 for each stream.
366The actual length of each missing Opus frame inside the packet is zero bytes,
367 as defined in Section&nbsp;3.2.1 of&nbsp;<xref target="RFC6716"/>.
368</t>
369
370<t>
371Zero-byte frames MAY be packed into packets using any of codes&nbsp;0, 1,
372 2, or&nbsp;3.
373When successive frames have the same configuration, the higher code packings
374 reduce overhead.
375Likewise, 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&nbsp;2 or VBR code&nbsp;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.
384Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth,
385 channel count, and frame size as the previous packet (if any).
386This 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&nbsp;4.5 of&nbsp;<xref target="RFC6716"/>.
389</t>
390
391<t>
392When 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.
395However, 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.
397Such changes SHOULD be delayed as long as possible to simplify
398 things for PLC implementations.
399</t>
400
401<t>
402As an example, a 95&nbsp;ms gap could be encoded as nineteen 5&nbsp;ms frames
403 in two bytes with a single CBR code&nbsp;3 packet.
404If the previous frame size was 20&nbsp;ms, using four 20&nbsp;ms frames
405 followed by three 5&nbsp;ms frames requires 4&nbsp;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.
408The total bitrate of the latter approach, including Ogg overhead, is about
409 0.4&nbsp;kbps, so the impact on file size is minimal.
410</t>
411
412<t>
413Changing modes is discouraged, since this causes some decoder implementations
414 to reset their PLC state.
415However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple
416 of 10&nbsp;ms.
417If 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>
422In the example above, if the previous frame was a 20&nbsp;ms SILK mode frame,
423 the better solution is to synthesize a packet describing four 20&nbsp;ms SILK
424 frames, followed by a packet with a single 10&nbsp;ms SILK
425 frame, and finally a packet with a 5&nbsp;ms CELT frame, to fill the 95&nbsp;ms
426 gap.
427This 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.
430At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality
431 solution.
432</t>
433
434<t>
435Since 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>
445There 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.
448The encoder might have introduced additional latency through its own resampling
449 and analysis (though the exact amount is not specified).
450Therefore, 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.
453These 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.
456However, a player will want to skip these samples after decoding them.
457</t>
458
459<t>
460A '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.
464This amount need not be a multiple of 2.5&nbsp;ms, MAY be smaller than a single
465 packet, or MAY span the contents of several packets.
466These samples are not valid audio.
467</t>
468
469<t>
470For example, if the first Opus frame uses the CELT mode, it will always
471 produce 120 samples of windowed overlap-add data.
472However, 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.
475The SILK mode requires additional delay to account for its analysis and
476 resampling latency.
477The encoder delays the original audio to avoid this problem.
478</t>
479
480<t>
481The pre-skip field MAY also be used to perform sample-accurate cropping of
482 already encoded streams.
483In this case, a value of at least 3840&nbsp;samples (80&nbsp;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>
492The 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>
502For 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>
507This 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>
518The initial PCM sample position before any samples are played is normally '0'.
519In 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>
527Vorbis 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"/>).
530However, 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.
533Opus 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>
542The 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.
545This is used to end the stream somewhere other than an even frame boundary.
546The 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.
549The difference between these granule positions indicates how many samples to
550 keep after decoding the packets that completed on the final page.
551The remaining samples are discarded.
552The 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>
560The 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.
564Allowing 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.
567This means that the PCM sample position just before the first sample to be
568 played MAY be larger than '0'.
569Synchronization when multiplexing with other logical streams still uses the PCM
570 sample position relative to '0' to compute sample times.
571This 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>
577On 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.
581An 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.
585It MAY defer this action until it decodes the last packet completed on that
586 page.
587</t>
588
589<t>
590If 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.
592This would indicate that there are more samples to be skipped from the initial
593 decoded output than exist in the stream.
594If 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.
598If 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.
602Both 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>
608Seeking 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.
611With appropriately weighted bisection, accurate seeking can be performed in
612 just one or two bisections on average, even in multi-gigabyte files.
613See <xref target="seeking"/> for an example of general implementation guidance.
614</t>
615
616<t>
617When seeking within an Ogg Opus stream, an implementation SHOULD start decoding
618 (and discarding the output) at least 3840&nbsp;samples (80&nbsp;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.
621This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the
622 beginning of the stream.
623If the point 80&nbsp;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&nbsp;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>
635An 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>
664The fields in the identification (ID) header have the following meaning:
665<list style="numbers">
666<t>Magic Signature:
667<vspace blankLines="1"/>
668This is an 8-octet (64-bit) field that allows codec identification and is
669 human-readable.
670It 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>
681Starting 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"/>
687The version number MUST always be '1' for this version of the encapsulation
688 specification.
689Implementations 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.
692That 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.
695For 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.
698The 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"/>
704This is the number of output channels.
705This might be different than the number of encoded channels, which can change
706 on a packet-by-packet basis.
707This value MUST NOT be zero.
708The maximum allowable value depends on the channel mapping family, and might be
709 as large as 255.
710See <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"/>
716This is the number of samples (at 48&nbsp;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.
719When cropping the beginning of existing Ogg Opus streams, a pre-skip of at
720 least 3,840&nbsp;samples (80&nbsp;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"/>
727This is the sample rate of the original input (before encoding), in Hz.
728This field is <spanx style="emph">not</spanx> the sample rate to use for
729 playback of the encoded data.
730<vspace blankLines="1"/>
731Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and
732 20&nbsp;kHz.
733Each packet in the stream can have a different audio bandwidth.
734Regardless of the audio bandwidth, the reference decoder supports decoding any
735 stream at a sample rate of 8, 12, 16, 24, or 48&nbsp;kHz.
736The original sample rate of the audio passed to the encoder is not preserved
737 by the lossy compression.
738<vspace blankLines="1"/>
739An 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&nbsp;kHz playback, decode at 48&nbsp;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&nbsp;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&nbsp;kHz and resample.</t>
749</list>
750However, the 'Input Sample Rate' field allows the muxer to pass the sample
751 rate of the original input stream as metadata.
752This is useful when the user requires the output sample rate to match the
753 input sample rate.
754For 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"/>
759A value of zero indicates 'unspecified'.
760Muxers 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).
764Implementations SHOULD support input sample rates between 8&nbsp;kHz and
765 192&nbsp;kHz (inclusive).
766Rates outside this range MAY be ignored by falling back to the default rate of
767 48&nbsp;kHz instead.
768<vspace blankLines="1"/>
769</t>
770<t>Output Gain (16 bits, signed, little endian):
771<vspace blankLines="1"/>
772This is a gain to be applied when decoding.
773It 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&nbsp;<xref target="q-notation"/>).
777<vspace blankLines="1"/>
778To apply the gain, an implementation could use
779<figure align="center">
780<artwork align="center"><![CDATA[
781sample *= 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"/>
786Players and media frameworks SHOULD apply it by default.
787If 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"/>
792A 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.
794A 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"/>
798Although 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.
801The 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"/>
808This octet indicates the order and semantic meaning of the output channels.
809<vspace blankLines="1"/>
810Each 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.
813The details are described in <xref target="channel_mapping"/>.
814</t>
815<t>Channel Mapping Table:
816This table defines the mapping from encoded streams to output channels.
817Its contents are specified in <xref target="channel_mapping"/>.
818</t>
819</list>
820</t>
821
822<t>
823All 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.
826Implementations 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.
829Future versions of this specification, even backwards-compatible versions,
830 might include additional fields in the ID header.
831If 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>
838An Ogg Opus stream allows mapping one number of Opus streams (N) to a possibly
839 larger number of decoded channels (M&nbsp;+&nbsp;N) to yet another number of
840 output channels (C), which might be larger or smaller than the number of
841 decoded channels.
842The 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&nbsp;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>
862The 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"/>
866This is the total number of streams encoded in each Ogg packet.
867This value is necessary to correctly parse the packed Opus packets inside an
868 Ogg packet, as described in <xref target="packet_organization"/>.
869This 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"/>
872For channel mapping family&nbsp;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):
876This is the number of streams whose decoders are to be configured to produce
877 two channels (stereo).
878This MUST be no larger than the total number of streams, N.
879<vspace blankLines="1"/>
880Each packet in an Opus stream has an internal channel count of 1 or 2, which
881 can change from packet to packet.
882This is selected by the encoder depending on the bitrate and the audio being
883 encoded.
884The original channel count of the audio passed to the encoder is not
885 necessarily preserved by the lossy compression.
886<vspace blankLines="1"/>
887Regardless 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.
890The '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&nbsp;-&nbsp;M) decoders are to be initialized for mono (a single
893 channel) only.
894The total number of decoded channels, (M&nbsp;+&nbsp;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"/>
898For channel mapping family&nbsp;0, this value defaults to (C&nbsp;-&nbsp;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):
903This contains one octet per output channel, indicating which decoded channel
904 is to be used for each one.
905Let 'index' be the value of this octet for a particular output channel.
906This value MUST either be smaller than (M&nbsp;+&nbsp;N), or be the special
907 value 255.
908If '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.
911If 'index' is 2*M or larger, but less than 255, the output MUST be taken from
912 decoding stream ('index'&nbsp;-&nbsp;M) as mono.
913If 'index' is 255, the corresponding output channel MUST contain pure silence.
914<vspace blankLines="1"/>
915The number of output channels, C, is not constrained to match the number of
916 decoded channels (M&nbsp;+&nbsp;N).
917A single index value MAY appear multiple times, i.e., the same decoded channel
918 might be mapped to multiple output channels.
919Some decoded channels might not be assigned to any output channel, as well.
920<vspace blankLines="1"/>
921For channel mapping family&nbsp;0, the first index defaults to 0, and if
922 C&nbsp;==&nbsp;2, the second index defaults to 1.
923Neither index is coded.
924</t>
925</list>
926</t>
927
928<t>
929After producing the output channels, the channel mapping family determines the
930 semantic meaning of each one.
931There are three defined mapping families in this specification.
932</t>
933
934<section anchor="channel_mapping_0" title="Channel Mapping Family 0">
935<t>
936Allowed numbers of channels: 1 or 2.
937RTP mapping.
938This 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>
945Special 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&nbsp;==&nbsp;2, with stream index&nbsp;0 mapped to output
948 channel&nbsp;0 (mono, or left channel) and stream index&nbsp;1 mapped to
949 output channel&nbsp;1 (right channel) if stereo.
950When 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>
957Allowed numbers of channels: 1...8.
958Vorbis channel order (see below).
959</t>
960<t>
961Each channel is assigned to a speaker location in a conventional surround
962 arrangement.
963Specific 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&nbsp;left, front&nbsp;right, rear&nbsp;left, rear&nbsp;right).</t>
970  <t>5 channels: 5.0 surround (front&nbsp;left, front&nbsp;center, front&nbsp;right, rear&nbsp;left, rear&nbsp;right).</t>
971  <t>6 channels: 5.1 surround (front&nbsp;left, front&nbsp;center, front&nbsp;right, rear&nbsp;left, rear&nbsp;right, LFE).</t>
972  <t>7 channels: 6.1 surround (front&nbsp;left, front&nbsp;center, front&nbsp;right, side&nbsp;left, side&nbsp;right, rear&nbsp;center, LFE).</t>
973  <t>8 channels: 7.1 surround (front&nbsp;left, front&nbsp;center, front&nbsp;right, side&nbsp;left, side&nbsp;right, rear&nbsp;left, rear&nbsp;right, LFE)</t>
974</list>
975</t>
976<t>
977This set of surround options and speaker location orderings is the same
978 as those used by the Vorbis codec <xref target="vorbis-mapping"/>.
979The 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.
986Implementations 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>
995Allowed numbers of channels: 1...255.
996No defined channel meaning.
997</t>
998<t>
999Channels are unidentified.
1000General-purpose players SHOULD NOT attempt to play these streams.
1001Offline implementations MAY deinterleave the output into separate PCM files,
1002 one per channel.
1003Implementations 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>
1012The remaining channel mapping families (2...254) are reserved.
1013A 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>
1020An 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.
1023Players SHOULD perform channel mixing to increase or reduce the number of
1024 channels as needed.
1025</t>
1026
1027<t>
1028Implementations MAY use the matrices in
1029 Figures&nbsp;<xref target="downmix-matrix-3" format="counter"/>
1030 through&nbsp;<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.
1034Matrices for 3 and 4 channels are normalized so each coefficient row sums
1035 to 1 to avoid clipping.
1036For 5 or more channels they are normalized to 2 as a compromise between
1037 clipping and dynamic range reduction.
1038</t>
1039<t>
1040In these matrices the front left and front right channels are generally
1041passed through directly.
1042When 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.
1045Rear 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[
1053L output = ( 0.585786 * left + 0.414214 * center                    )
1054R output = (                   0.414214 * center + 0.585786 * right )
1055]]></artwork>
1056<postamble>
1057Exact coefficient values are 1 and 1/sqrt(2), multiplied by
1058 1/(1&nbsp;+&nbsp;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>
1072Exact coefficient values are 1, sqrt(3)/2 and 1/2, multiplied by
1073 1/(1&nbsp;+&nbsp;sqrt(3)/2&nbsp;+&nbsp;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>
1089Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
1090 2/(1&nbsp;+&nbsp;1/sqrt(2)&nbsp;+&nbsp;sqrt(3)/2&nbsp;+&nbsp;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>
1107Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
11082/(1&nbsp;+&nbsp;1/sqrt(2)&nbsp;+&nbsp;sqrt(3)/2&nbsp;+&nbsp;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>
1123Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2, 1/2 and
1124 sqrt(3)/2/sqrt(2), multiplied by
1125 2/(1&nbsp;+&nbsp;1/sqrt(2)&nbsp;+&nbsp;sqrt(3)/2&nbsp;+&nbsp;1/2 +
1126 sqrt(3)/2/sqrt(2) + 1/sqrt(2)) for normalization.
1127The 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>
1142Exact coefficient values are 1, 1/sqrt(2), sqrt(3)/2 and 1/2, multiplied by
1143 2/(2&nbsp;+&nbsp;2/sqrt(2)&nbsp;+&nbsp;sqrt(3)) for normalization.
1144The 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>
1188The 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"/>
1195This is an 8-octet (64-bit) field that allows codec identification and is
1196 human-readable.
1197It 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>
1208Starting 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"/>
1214This field gives the length of the following vendor string, in octets.
1215It 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"/>
1221This is a simple human-readable tag for vendor information, encoded as a UTF-8
1222 string&nbsp;<xref target="RFC3629"/>.
1223No terminating null octet is necessary.
1224<vspace blankLines="1"/>
1225This tag is intended to identify the codec encoder and encapsulation
1226 implementations, for tracing differences in technical behavior.
1227User-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"/>
1233This field indicates the number of user-supplied comments.
1234It MAY indicate there are zero user-supplied comments, in which case there are
1235 no additional fields in the packet.
1236It 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"/>
1242This field gives the length of the following user comment string, in octets.
1243There is one for each user comment indicated by the 'user comment list length'
1244 field.
1245It 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"/>
1250This field contains a single user comment encoded as a UTF-8
1251 string&nbsp;<xref target="RFC3629"/>.
1252There is one for each user comment indicated by the 'user comment list length'
1253 field.
1254</t>
1255</list>
1256</t>
1257
1258<t>
1259The 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.
1264Making 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>
1271Immediately following the user comment list, the comment header MAY
1272 contain zero-padding or other binary data which is not specified here.
1273If 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.
1277This allows informal experimentation with the format of this binary data until
1278 it can be specified later.
1279</t>
1280
1281<t>
1282The comment header can be arbitrarily large and might be spread over a large
1283 number of Ogg pages.
1284Implementations MUST avoid attempting to allocate excessive amounts of memory
1285 when presented with a very large comment header.
1286To accomplish this, implementations MAY treat a stream as invalid if it has a
1287 comment header larger than 125,829,120&nbsp;octets (120&nbsp;MB), and MAY
1288 ignore individual comments that are not fully contained within the first
1289 61,440&nbsp;octets of the comment header.
1290</t>
1291
1292<section anchor="comment_format" title="Tag Definitions">
1293<t>
1294The 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>
1299Two 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[
1305R128_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.
1311The gain is a Q7.8 fixed point number in dB, as in the ID header's 'output
1312 gain' field.
1313This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in
1314 Vorbis&nbsp;<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[
1320R128_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.
1326The gain is also a Q7.8 fixed point number in dB, as in the ID header's
1327 'output gain' field.
1328The values '-573' and '111' given here are just examples.
1329</t>
1330<t>
1331An 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.
1334A leading '+' or '-' character is valid.
1335Leading zeros are also permitted, but the value MUST be represented by
1336 no more than 6 characters.
1337Other non-digit characters MUST NOT be present.
1338</t>
1339<t>
1340If 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.
1343If 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.
1346If 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.
1348A 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>
1352To 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.
1358Peak normalizations are difficult to calculate reliably for lossy codecs
1359 because of variation in excursion heights due to decoder differences.
1360In 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>
1370Technically, 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.
1372These packets might be spread over a similarly enormous number of Ogg pages.
1373When 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.
1377Demuxers 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&nbsp;octets per Opus stream, unless they have a specific reason for
1380 allowing extra padding.
1381Such packets necessarily contain more padding than needed to make a stream CBR.
1382Demuxers MUST avoid attempting to allocate excessive amounts of memory when
1383 presented with a very large packet.
1384Demuxers MAY treat audio data packets as invalid or partially process them if
1385 they are larger than 61,440&nbsp;octets in an Ogg Opus stream with channel
1386 mapping families&nbsp;0 or&nbsp;1.
1387Demuxers 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&nbsp;octets and also
1389 larger than 7,680&nbsp;octets per Opus stream.
1390The presence of an extremely large packet in the stream could indicate a
1391 memory exhaustion attack or stream corruption.
1392</t>
1393<t>
1394In an Ogg Opus stream, the largest possible valid packet that does not use
1395 padding has a size of (61,298*N&nbsp;-&nbsp;2) octets.
1396With 255&nbsp;streams, this is 15,630,988&nbsp;octets and can
1397 span up to 61,298&nbsp;Ogg pages, all but one of which will have a granule
1398 position of -1.
1399This is of course a very extreme packet, consisting of 255&nbsp;streams, each
1400 containing 120&nbsp;ms of audio encoded as 2.5&nbsp;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&nbsp;3 Opus packet).
1403Even in such a packet, most of the data will be zeros as 2.5&nbsp;ms frames
1404 cannot actually use all 1275&nbsp;octets.
1405</t>
1406<t>
1407The largest packet consisting of entirely useful data is
1408 (15,326*N&nbsp;-&nbsp;2) octets.
1409This corresponds to 120&nbsp;ms of audio encoded as 10&nbsp;ms frames in either
1410 SILK or Hybrid mode, but at a data rate of over 1&nbsp;Mbps, which makes little
1411 sense for the quality achieved.
1412</t>
1413<t>
1414A more reasonable limit is (7,664*N&nbsp;-&nbsp;2) octets.
1415This corresponds to 120&nbsp;ms of audio encoded as 20&nbsp;ms stereo CELT mode
1416 frames, with a total bitrate just under 511&nbsp;kbps (not counting the Ogg
1417 encapsulation overhead).
1418For 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.
1421This gives a size of 61,310&nbsp;octets, which is rounded up to a multiple of
1422 1,024&nbsp;octets to yield the audio data packet size of 61,440&nbsp;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>
1429When encoding Opus streams, Ogg muxers SHOULD take into account the
1430 algorithmic delay of the Opus encoder.
1431</t>
1432<t>
1433In encoders derived from the reference
1434 implementation&nbsp;<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>
1443To 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.
1447For more information on linear prediction, see
1448 <xref target="linear-prediction"/>.
1449For an input file containing 'length' samples, the implementation SHOULD set
1450 the pre-skip header value to (delay_samples&nbsp;+&nbsp;extra_samples), encode
1451 at least (length&nbsp;+&nbsp;delay_samples&nbsp;+&nbsp;extra_samples)
1452 samples, and set the granule position of the last page to
1453 (length&nbsp;+&nbsp;delay_samples&nbsp;+&nbsp;extra_samples).
1454This 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>
1461The first step in LPC extrapolation is to compute linear prediction
1462 coefficients. <xref target="lpc-sample"/>
1463When 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.
1465The last N samples are used as memory to an infinite impulse response (IIR)
1466 filter.
1467</t>
1468<t>
1469The filter is then applied on a zero input to extrapolate the end of the signal.
1470Let 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       ---
1477x(n) = \   a(k)*x(n-k)
1478       /
1479       ---
1480       k=1
1481]]></artwork>
1482</figure>
1483<t>
1484The process is repeated independently for each channel.
1485It is possible to extend the beginning of the signal by applying the same
1486 process backward in time.
1487When 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>
1496In 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.
1498This can be done simply by separating the input streams into segments and
1499 encoding each segment independently.
1500The drawback of this approach is that it creates a small discontinuity
1501 at the boundary due to the lossy nature of Opus.
1502A 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.
1506De-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>
1518In 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>
1527For 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>
1537A 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>
1550Implementations of the Opus codec need to take appropriate security
1551 considerations into account, as outlined in <xref target="RFC4732"/>.
1552This is just as much a problem for the container as it is for the codec itself.
1553Malicious payloads and/or input streams can be used to attack codec
1554 implementations.
1555Implementations MUST NOT overrun their allocated memory nor consume excessive
1556 resources when decoding payloads or processing input streams.
1557Although 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>
1563Header parsing code contains the most likely area for potential overruns.
1564It 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).
1567Implementations 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>
1574To 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.
1577For 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.
1580Even 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.
1583After 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&nbsp;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.
1588If allocating an array of pointers to point at these strings, the size of the
1589 pointers may be larger than 4&nbsp;octets, potentially requiring a separate
1590 overflow check.
1591</t>
1592
1593<t>
1594Another bug in this class we have observed more than once involves the handling
1595 of invalid data at the end of a stream.
1596Often, implementations will seek to the end of a stream to locate the last
1597 timestamp in order to compute its total duration.
1598If they do not find a valid capture pattern and Ogg page from the desired
1599 logical stream, they will back up and try again.
1600If 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>
1606In 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>
1613Like 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.
1616Elements 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>
1624An "Ogg Opus file" consists of one or more sequentially multiplexed segments,
1625 each containing exactly one Ogg Opus stream.
1626The RECOMMENDED mime-type for Ogg Opus files is "audio/ogg".
1627</t>
1628
1629<t>
1630If 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>
1644The RECOMMENDED filename extension for Ogg Opus files is '.opus'.
1645</t>
1646
1647<t>
1648When 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"/>.
1651Such 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.
1654In such cases the the '.opus' filename extension is NOT RECOMMENDED.
1655</t>
1656
1657<t>
1658In 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>
1666This 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>
1672This 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.
1675IANA is requested to create a new name space of "Opus Channel Mapping
1676 Families".
1677This will be a new registry on the IANA Matrix, and not a subregistry of an
1678 existing registry.
1679Modifications to this registry follow the "Specification Required" registration
1680 policy as defined in <xref target="RFC5226"/>.
1681Each 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)
1684Each 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>
1691This 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>
1700The 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&nbsp;<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>
1710Thanks 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.
1713Additional 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>
1720In&nbsp;<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 &quot;Monty&quot; 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&amp;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&amp;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&amp;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 &quot;Monty&quot; 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&nbsp;A: Embedding Vorbis
1852      into an Ogg stream</title>
1853    <author initials="C." surname="Montgomery"
1854     fullname="Christopher &quot;Monty&quot; 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