doc/draft-ietf-codec-oggopus.xml - Issue 882843002: Update to opus-HEAD-66611f1.

Unified Diff: doc/draft-ietf-codec-oggopus.xml

Issue 882843002: Update to opus-HEAD-66611f1. (Closed) Base URL: https://chromium.googlesource.com/chromium/deps/opus.git@master

Patch Set: Created 5 years, 11 months ago

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Index: doc/draft-ietf-codec-oggopus.xml

diff --git a/doc/draft-ietf-codec-oggopus.xml b/doc/draft-ietf-codec-oggopus.xml

index 6131e69ed5aaa8179ee6fe6b4b84a747cd26ce3f..e5181c92ebc6ad0454cfaa9874abfd90403c8ca0 100644

--- a/doc/draft-ietf-codec-oggopus.xml

+++ b/doc/draft-ietf-codec-oggopus.xml

@@ -1,16 +1,17 @@

<?xml version="1.0" encoding="utf-8"?>

<!DOCTYPE rfc SYSTEM 'rfc2629.dtd' [

-<!ENTITY rfc2119 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.2119.xml'>

-<!ENTITY rfc3533 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.3533.xml'>

-<!ENTITY rfc3629 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.3629.xml'>

-<!ENTITY rfc4732 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.4732.xml'>

-<!ENTITY rfc5334 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.5334.xml'>

-<!ENTITY rfc6381 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.6381.xml'>

-<!ENTITY rfc6716 PUBLIC '' 'https://xml2rfc.tools.ietf.org/tools/xml2rfc/public/rfc/bibxml/reference.RFC.6716.xml'>

+<!ENTITY rfc2119 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml'>

+<!ENTITY rfc3533 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3533.xml'>

+<!ENTITY rfc3629 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.3629.xml'>

+<!ENTITY rfc4732 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.4732.xml'>

+<!ENTITY rfc5334 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.5334.xml'>

+<!ENTITY rfc6381 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6381.xml'>

+<!ENTITY rfc6716 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6716.xml'>

+<!ENTITY rfc6982 PUBLIC '' 'http://xml.resource.org/public/rfc/bibxml/reference.RFC.6982.xml'>

<?rfc toc="yes" symrefs="yes" ?>

-<rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-01">

+<rfc ipr="trust200902" category="std" docName="draft-ietf-codec-oggopus-06">

<front>

<title abbrev="Ogg Opus">Ogg Encapsulation for the Opus Audio Codec</title>

@@ -54,12 +55,12 @@

<country>Canada</country>

</postal>

-<phone>+1 604 778 1540</phone>

+<phone>+1 778 785 1540</phone>

<email>giles@xiph.org</email>

</address>

</author>

-<date day="24" month="May" year="2013"/>

+<date day="18" month="October" year="2014"/>

<workgroup>codec</workgroup>

@@ -100,7 +101,7 @@ Each page is associated with a particular logical stream and contains a capture

stream, to aid seeking.

A single page can contain up to 65,025 octets of packet data from up to 255

different packets.

-Packets may be split arbitrarily across pages, and continued from one page to

+Packets MAY be split arbitrarily across pages, and continued from one page to

the next (allowing packets much larger than would fit on a single page).

Each page contains 'lacing values' that indicate how the data is partitioned

into packets, allowing a demuxer to recover the packet boundaries without

@@ -109,7 +110,7 @@ A packet is said to 'complete' on a page when the page contains the final

lacing value corresponding to that packet.

</t>

<t>

-This encapsulation defines the required contents of the packet data, including

+This encapsulation defines the contents of the packet data, including

the necessary headers, the organization of those packets into a logical

stream, and the interpretation of the codec-specific granule position field.

It does not attempt to describe or specify the existing Ogg container format.

@@ -122,8 +123,8 @@ Readers unfamiliar with the basic concepts mentioned above are encouraged to

<t>

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",

- "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be

- interpreted as described in <xref target="RFC2119"/>.

+ "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this

+ document are to be interpreted as described in <xref target="RFC2119"/>.

</t>

<t>

@@ -138,7 +139,7 @@ All other implementations are "unconditionally compliant".

<t>

-An Opus stream is organized as follows.

+An Ogg Opus stream is organized as follows.

</t>

<t>

There are two mandatory header packets.

@@ -149,7 +150,7 @@ The first packet in the logical Ogg bitstream MUST contain the identification

(ID) header, which uniquely identifies a stream as Opus audio.

The format of this header is defined in <xref target="id_header"/>.

It MUST be placed alone (without any other packet data) on the first page of

- the logical Ogg bitstream, and must complete on that page.

+ the logical Ogg bitstream, and MUST complete on that page.

This page MUST have its 'beginning of stream' flag set.

</t>

<t>

@@ -165,8 +166,8 @@ However many pages it spans, the comment header packet MUST finish the page on

All subsequent pages are audio data pages, and the Ogg packets they contain are

audio data packets.

Each audio data packet contains one Opus packet for each of N different

- streams, where N is typically one for mono or stereo, but may be greater than

- one for, e.g., multichannel audio.

+ streams, where N is typically one for mono or stereo, but MAY be greater than

+ one for multichannel audio.

The value N is specified in the ID header (see

<xref target="channel_mapping"/>), and is fixed over the entire length of the

logical Ogg bitstream.

@@ -179,11 +180,17 @@ The remaining Opus packet is packed at the end of the Ogg packet using the

regular, undelimited framing from Section 3 of <xref target="RFC6716"/>.

All of the Opus packets in a single Ogg packet MUST be constrained to have the

same duration.

-The duration and coding modes of each Opus packet are contained in the

- TOC (table of contents) sequence in the first few bytes.

A decoder SHOULD treat any Opus packet whose duration is different from that of

- the first Opus packet in an Ogg packet as if it were an Opus packet with an

- illegal TOC sequence.

+ the first Opus packet in an Ogg packet as if it were a malformed Opus packet

+ with an invalid TOC sequence.

+</t>

+<t>

+The coding mode (SILK, Hybrid, or CELT), audio bandwidth, channel count,

+ duration (frame size), and number of frames per packet, are indicated in the

+ TOC (table of contents) sequence at the beginning of each Opus packet, as

+ described in Section 3.1 of <xref target="RFC6716"/>.

+The combination of mode, audio bandwidth, and frame size is referred to as

+ the configuration of an Opus packet.

</t>

<t>

The first audio data page SHOULD NOT have the 'continued packet' flag set

@@ -191,13 +198,15 @@ The first audio data page SHOULD NOT have the 'continued packet' flag set

page).

Packets MUST be placed into Ogg pages in order until the end of stream.

Audio packets MAY span page boundaries.

-A decoder MUST treat a zero-octet audio data packet as if it were an Opus

- packet with an illegal TOC sequence.

+A decoder MUST treat a zero-octet audio data packet as if it were a malformed

+ Opus packet as described in Section 3.4 of <xref target="RFC6716"/>.

+</t>

+<t>

The last page SHOULD have the 'end of stream' flag set, but implementations

- should be prepared to deal with truncated streams that do not have a page

+ need to be prepared to deal with truncated streams that do not have a page

marked 'end of stream'.

The final packet on the last page SHOULD NOT be a continued packet, i.e., the

- final lacing value should be less than 255.

+ final lacing value SHOULD be less than 255.

There MUST NOT be any more pages in an Opus logical bitstream after a page

marked 'end of stream'.

</t>

@@ -223,7 +232,7 @@ It is possible to run an Opus decoder at other sampling rates, but the value

</t>

<t>

-The duration of an Opus packet may be any multiple of 2.5 ms, up to a

+The duration of an Opus packet can be any multiple of 2.5 ms, up to a

maximum of 120 ms.

This duration is encoded in the TOC sequence at the beginning of each packet.

The number of samples returned by a decoder corresponds to this duration

@@ -245,19 +254,105 @@ All other pages with completed packets after the first MUST have a granule

This guarantees that a demuxer can assign individual packets the same granule

position when working forwards as when working backwards.

For this to work, there cannot be any gaps.

-In order to support capturing a stream that uses discontinuous transmission

- (DTX), an encoder SHOULD emit packets that explicitly request the use of

- Packet Loss Concealment (PLC) (i.e., with a frame length of 0, as defined in

- Section 3.2.1 of <xref target="RFC6716"/>) in place of the packets that were

- not transmitted.

</t>

+<section anchor="gap-repair" title="Repairing Gaps in Real-time Streams">

+<t>

+In order to support capturing a real-time stream that has lost or not

+ transmitted packets, a muxer SHOULD emit packets that explicitly request the

+ use of Packet Loss Concealment (PLC) in place of the missing packets.

+Only gaps that are a multiple of 2.5 ms are repairable, as these are the

+ only durations that can be created by packet loss or discontinuous

+ transmission.

+Muxers need not handle other gap sizes.

+Creating the necessary packets involves synthesizing a TOC byte (defined in

+Section 3.1 of <xref target="RFC6716"/>)—and whatever

+ additional internal framing is needed—to indicate the packet duration

+ for each stream.

+The actual length of each missing Opus frame inside the packet is zero bytes,

+ as defined in Section 3.2.1 of <xref target="RFC6716"/>.

+</t>

+<t>

+Zero-byte frames MAY be packed into packets using any of codes 0, 1,

+ 2, or 3.

+When successive frames have the same configuration, the higher code packings

+ reduce overhead.

+Likewise, if the TOC configuration matches, the muxer MAY further combine the

+ empty frames with previous or subsequent non-zero-length frames (using

+ code 2 or VBR code 3).

+</t>

+<t>

+<xref target="RFC6716"/> does not impose any requirements on the PLC, but this

+ section outlines choices that are expected to have a positive influence on

+ most PLC implementations, including the reference implementation.

+Synthesized TOC sequences SHOULD maintain the same mode, audio bandwidth,

+ channel count, and frame size as the previous packet (if any).

+This is the simplest and usually the most well-tested case for the PLC to

+ handle and it covers all losses that do not include a configuration switch,

+ as defined in Section 4.5 of <xref target="RFC6716"/>.

+</t>

+<t>

+When a previous packet is available, keeping the audio bandwidth and channel

+ count the same allows the PLC to provide maximum continuity in the concealment

+ data it generates.

+However, if the size of the gap is not a multiple of the most recent frame

+ size, then the frame size will have to change for at least some frames.

+Such changes SHOULD be delayed as long as possible to simplify

+ things for PLC implementations.

+</t>

+<t>

+As an example, a 95 ms gap could be encoded as nineteen 5 ms frames

+ in two bytes with a single CBR code 3 packet.

+If the previous frame size was 20 ms, using four 20 ms frames

+ followed by three 5 ms frames requires 4 bytes (plus an extra byte

+ of Ogg lacing overhead), but allows the PLC to use its well-tested steady

+ state behavior for as long as possible.

+The total bitrate of the latter approach, including Ogg overhead, is about

+ 0.4 kbps, so the impact on file size is minimal.

+</t>

+<t>

+Changing modes is discouraged, since this causes some decoder implementations

+ to reset their PLC state.

+However, SILK and Hybrid mode frames cannot fill gaps that are not a multiple

+ of 10 ms.

+If switching to CELT mode is needed to match the gap size, a muxer SHOULD do

+ so at the end of the gap to allow the PLC to function for as long as possible.

+</t>

+<t>

+In the example above, if the previous frame was a 20 ms SILK mode frame,

+ the better solution is to synthesize a packet describing four 20 ms SILK

+ frames, followed by a packet with a single 10 ms SILK

+ frame, and finally a packet with a 5 ms CELT frame, to fill the 95 ms

+ gap.

+This also requires four bytes to describe the synthesized packet data (two

+ bytes for a CBR code 3 and one byte each for two code 0 packets) but three

+ bytes of Ogg lacing overhead are needed to mark the packet boundaries.

+At 0.6 kbps, this is still a minimal bitrate impact over a naive, low quality

+ solution.

+</t>

+<t>

+Since medium-band audio is an option only in the SILK mode, wideband frames

+ SHOULD be generated if switching from that configuration to CELT mode, to

+ ensure that any PLC implementation which does try to migrate state between

+ the modes will be able to preserve all of the available audio bandwidth.

+</t>

+</section>

<t>

There is some amount of latency introduced during the decoding process, to

- allow for overlap in the MDCT modes, stereo mixing in the LP modes, and

- resampling, and the encoder will introduce even more latency (though the exact

- amount is not specified).

+ allow for overlap in the CELT mode, stereo mixing in the SILK mode, and

+ resampling.

+The encoder might have introduced additional latency through its own resampling

+ and analysis (though the exact amount is not specified).

Therefore, the first few samples produced by the decoder do not correspond to

real input audio, but are instead composed of padding inserted by the encoder

to compensate for this latency.

@@ -271,20 +366,39 @@ However, a decoder will want to skip these samples after decoding them.

A 'pre-skip' field in the ID header (see <xref target="id_header"/>) signals

the number of samples which SHOULD be skipped (decoded but discarded) at the

beginning of the stream.

-This provides sufficient history to the decoder so that it has already

- converged before the stream's output begins.

-It may also be used to perform sample-accurate cropping of existing encoded

- streams.

-This amount need not be a multiple of 2.5 ms, may be smaller than a single

- packet, or may span the contents of several packets.

+This amount need not be a multiple of 2.5 ms, MAY be smaller than a single

+ packet, or MAY span the contents of several packets.

+These samples are not valid audio, and SHOULD NOT be played.

</t>

+<t>

+For example, if the first Opus frame uses the CELT mode, it will always

+ produce 120 samples of windowed overlap-add data.

+However, the overlap data is initially all zeros (since there is no prior

+ frame), meaning this cannot, in general, accurately represent the original

+ audio.

+The SILK mode requires additional delay to account for its analysis and

+ resampling latency.

+The encoder delays the original audio to avoid this problem.

+</t>

+<t>

+The pre-skip field MAY also be used to perform sample-accurate cropping of

+ already encoded streams.

+In this case, a value of at least 3840 samples (80 ms) provides

+ sufficient history to the decoder that it will have converged

+ before the stream's output begins.

+</t>

</section>

<t>

+<figure align="center">

+<preamble>

The PCM sample position is determined from the granule position using the

formula

-<figure align="center">

+</preamble>

<artwork align="center"><![CDATA[

'PCM sample position' = 'granule position' - 'pre-skip' .

]]></artwork>

@@ -295,8 +409,10 @@ The PCM sample position is determined from the granule position using the

For example, if the granule position of the first audio data page is 59,971,

and the pre-skip is 11,971, then the PCM sample position of the last decoded

sample from that page is 48,000.

-This can be converted into a playback time using the formula

+<preamble>

+This can be converted into a playback time using the formula

+</preamble>

<artwork align="center"><![CDATA[

'PCM sample position'

'playback time' = --------------------- .

@@ -317,12 +433,12 @@ In this case, the PCM sample position of the first audio sample to be played

<t>

Vorbis streams use a granule position smaller than the number of audio samples

contained in the first audio data page to indicate that some of those samples

- must be trimmed from the output (see <xref target="vorbis-trim"/>).

+ are trimmed from the output (see <xref target="vorbis-trim"/>).

However, to do so, Vorbis requires that the first audio data page contains

exactly two packets, in order to allow the decoder to perform PCM position

adjustments before needing to return any PCM data.

Opus uses the pre-skip mechanism for this purpose instead, since the encoder

- may introduce more than a single packet's worth of latency, and since very

+ MAY introduce more than a single packet's worth of latency, and since very

large packets in streams with a very large number of channels might not fit

on a single page.

</t>

@@ -356,11 +472,11 @@ Allowing a granule position larger than the number of samples allows the

beginning of a stream to be cropped or a live stream to be joined without

rewriting the granule position of all the remaining pages.

This means that the PCM sample position just before the first sample to be

- played may be larger than '0'.

+ played MAY be larger than '0'.

Synchronization when multiplexing with other logical streams still uses the PCM

sample position relative to '0' to compute sample times.

This does not affect the behavior of pre-skip: exactly 'pre-skip' samples

- should be skipped from the beginning of the decoded output, even if the

+ SHOULD be skipped from the beginning of the decoded output, even if the

initial PCM sample position is greater than zero.

</t>

@@ -368,7 +484,7 @@ This does not affect the behavior of pre-skip: exactly 'pre-skip' samples

On the other hand, a granule position that is smaller than the number of

decoded samples prevents a demuxer from working backwards to assign each

packet or each individual sample a valid granule position, since granule

- positions must be non-negative.

+ positions are non-negative.

A decoder MUST reject as invalid any stream where the granule position is

smaller than the number of samples contained in packets that complete on the

first audio data page with a completed packet, unless that page has the 'end

@@ -380,7 +496,7 @@ It MAY defer this action until it decodes the last packet completed on that

<t>

If that page has the 'end of stream' flag set, a demuxer MUST reject as invalid

any stream where its granule position is smaller than the 'pre-skip' amount.

-This would indicate that more samples should be skipped from the initial

+This would indicate that there are more samples to be skipped from the initial

decoded output than exist in the stream.

If the granule position is smaller than the number of decoded samples produced

by the packets that complete on that page, then a demuxer MUST use an initial

@@ -414,8 +530,8 @@ This 'pre-roll' is separate from, and unrelated to, the 'pre-skip' used at the

If the point 80 ms prior to the seek target comes before the initial PCM

sample position, the decoder SHOULD start decoding from the beginning of the

stream, applying pre-skip as normal, regardless of whether the pre-skip is

- larger or smaller than 80 ms, and then continue to discard the samples

- required to reach the seek target (if any).

+ larger or smaller than 80 ms, and then continue to discard samples

+ to reach the seek target (if any).

</t>

</section>

@@ -518,9 +634,9 @@ When cropping the beginning of existing Ogg Opus streams, a pre-skip of at

This field is <spanx style="emph">not</spanx> the sample rate to use for

playback of the encoded data.

-Opus has a handful of coding modes, with internal audio bandwidths of 4, 6, 8,

- 12, and 20 kHz.

-Each packet in the stream may have a different audio bandwidth.

+Opus can switch between internal audio bandwidths of 4, 6, 8, 12, and

+ 20 kHz.

+Each packet in the stream can have a different audio bandwidth.

Regardless of the audio bandwidth, the reference decoder supports decoding any

stream at a sample rate of 8, 12, 16, 24, or 48 kHz.

The original sample rate of the encoder input is not preserved by the lossy

@@ -533,12 +649,13 @@ An Ogg Opus player SHOULD select the playback sample rate according to the

<t>Otherwise, if the hardware's highest available sample rate is a supported

rate, decode at this sample rate.</t>

<t>Otherwise, if the hardware's highest available sample rate is less than

- 48 kHz, decode at the highest supported rate above this and resample.</t>

+ 48 kHz, decode at the next highest supported rate above this and

+ resample.</t>

<t>Otherwise, decode at 48 kHz and resample.</t>

</list>

However, the 'Input Sample Rate' field allows the encoder to pass the sample

rate of the original input stream as metadata.

-This may be useful when the user requires the output sample rate to match the

+This is useful when the user requires the output sample rate to match the

input sample rate.

For example, a non-player decoder writing PCM format samples to disk might

choose to resample the output audio back to the original input sample rate to

@@ -559,39 +676,42 @@ This is a gain to be applied by the decoder.

It is 20*log10 of the factor to scale the decoder output by to achieve the

desired playback volume, stored in a 16-bit, signed, two's complement

fixed-point value with 8 fractional bits (i.e., Q7.8).

-To apply the gain, a decoder could use

+<preamble>

+To apply the gain, a decoder could use

+</preamble>

<artwork align="center"><![CDATA[

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

]]></artwork>

-</figure>

+<postamble>

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

+</postamble>

+</figure>

-Virtually all players and media frameworks should apply it by default.

+Virtually all players and media frameworks SHOULD apply it by default.

If a player chooses to apply any volume adjustment or gain modification, such

- as the R128_TRACK_GAIN (see <xref target="comment_header"/>) or a user-facing

- volume knob, the adjustment MUST be applied in addition to this output gain in

- order to achieve playback at the desired volume.

+ as the R128_TRACK_GAIN (see <xref target="comment_header"/>), the adjustment

+ MUST be applied in addition to this output gain in order to achieve playback

+ at the normalized volume.

An encoder SHOULD set this field to zero, and instead apply any gain prior to

encoding, when this is possible and does not conflict with the user's wishes.

-The output gain should only be nonzero when the gain is adjusted after

- encoding, or when the user wishes to adjust the gain for playback while

- preserving the ability to recover the original signal amplitude.

+A nonzero output gain indicates the gain was adjusted after encoding, or that

+ a user wished to adjust the gain for playback while preserving the ability

+ to recover the original signal amplitude.

Although the output gain has enormous range (+/- 128 dB, enough to amplify

inaudible sounds to the threshold of physical pain), most applications can

only reasonably use a small portion of this range around zero.

The large range serves in part to ensure that gain can always be losslessly

- transferred between OpusHead and R128_TRACK_GAIN (see below) without

+ transferred between OpusHead and R128 gain tags (see below) without

saturating.

</t>

<t><spanx style="strong">Channel Mapping Family</spanx> (8 bits,

unsigned):

-This octet indicates the order and semantic meaning of the various channels

- encoded in each Ogg packet.

+This octet indicates the order and semantic meaning of the output channels.

Each possible value of this octet indicates a mapping family, which defines a

set of allowed channel counts, and the ordered set of channel names for each

@@ -651,7 +771,7 @@ The fields in the channel mapping table have the following meaning:

<t><spanx style="strong">Stream Count</spanx> 'N' (8 bits, unsigned):

This is the total number of streams encoded in each Ogg packet.

-This value is required to correctly parse the packed Opus packets inside an

+This value is necessary to correctly parse the packed Opus packets inside an

Ogg packet, as described in <xref target="packet_organization"/>.

This value MUST NOT be zero, as without at least one Opus packet with a valid

TOC sequence, a demuxer cannot recover the duration of an Ogg packet.

@@ -660,7 +780,7 @@ For channel mapping family 0, this value defaults to 1, and is not coded.

</t>

<t><spanx style="strong">Coupled Stream Count</spanx> 'M' (8 bits, unsigned):

-This is the number of streams whose decoders should be configured to produce

+This is the number of streams whose decoders are to be configured to produce

two channels.

This MUST be no larger than the total number of streams, N.

@@ -675,8 +795,8 @@ Regardless of the internal channel count, any Opus stream can be decoded as

mono (a single channel) or stereo (two channels) by appropriate initialization

of the decoder.

The 'coupled stream count' field indicates that the first M Opus decoders are

- to be initialized in stereo mode, and the remaining N-M decoders are to be

- initialized in mono mode.

+ to be initialized for stereo output, and the remaining N-M decoders are to be

+ initialized for mono only.

The total number of decoded channels, (M+N), MUST be no larger than 255, as

there is no way to index more channels than that in the channel mapping.

@@ -686,14 +806,14 @@ For channel mapping family 0, this value defaults to C-1 (i.e., 0 for mono

</t>

<t><spanx style="strong">Channel Mapping</spanx> (8*C bits):

This contains one octet per output channel, indicating which decoded channel

- should be used for each one.

+ is to be used for each one.

Let 'index' be the value of this octet for a particular output channel.

This value MUST either be smaller than (M+N), or be the special value 255.

If 'index' is less than 2*M, the output MUST be taken from decoding stream

('index'/2) as stereo and selecting the left channel if 'index' is even, and

the right channel if 'index' is odd.

-If 'index' is 2*M or larger, the output MUST be taken from decoding stream

- ('index'-M) as mono.

+If 'index' is 2*M or larger, but less than 255, the output MUST be taken from

+ decoding stream ('index'-M) as mono.

If 'index' is 255, the corresponding output channel MUST contain pure silence.

The number of output channels, C, is not constrained to match the number of

@@ -712,7 +832,7 @@ Neither index is coded.

<t>

After producing the output channels, the channel mapping family determines the

semantic meaning of each one.

-Currently there are three defined mapping families, although more may be added.

+There are three defined mapping families in this specification.

</t>

@@ -742,7 +862,7 @@ Vorbis channel order.

</t>

<t>

Each channel is assigned to a speaker location in a conventional surround

- configuration.

+ arrangement.

Specific locations depend on the number of channels, and are given below

in order of the corresponding channel indicies.

@@ -755,19 +875,20 @@ Specific locations depend on the number of channels, and are given below

<t>7 channels: 6.1 surround (front left, front center, front right, side left, side right, rear center, LFE).</t>

<t>8 channels: 7.1 surround (front left, front center, front right, side left, side right, rear left, rear right, LFE)</t>

</list>

-This set of surround configurations and speaker location orderings is the same

- as the one used by the Vorbis codec <xref target="vorbis-mapping"/>.

+</t>

+<t>

+This set of surround options and speaker location orderings is the same

+ as those used by the Vorbis codec <xref target="vorbis-mapping"/>.

The ordering is different from the one used by the

WAVE <xref target="wave-multichannel"/> and

FLAC <xref target="flac"/> formats,

- so correct ordering requires permutation of the output channels when encoding

- from or decoding to those formats.

+ so correct ordering requires permutation of the output channels when decoding

+ to or encoding from those formats.

'LFE' here refers to a Low Frequency Effects, often mapped to a subwoofer

- with no particular spacial position.

+ with no particular spatial position.

Implementations SHOULD identify 'side' or 'rear' speaker locations with

'surround' and 'back' as appropriate when interfacing with audio formats

or systems which prefer that terminology.

-Speaker configurations other than those described here are not supported.

</t>

</section>

@@ -811,7 +932,7 @@ Implementations MAY use the following matricies to implement downmixing from

Family 1</xref>, which are known to give acceptable results for stereo.

Matricies for 3 and 4 channels are normalized so each coefficent row sums

to 1 to avoid clipping.

-For 5 or more channels they are normalized to 2 as a compromize between

+For 5 or more channels they are normalized to 2 as a compromise between

clipping and dynamic range reduction.

</t>

<t>

@@ -828,8 +949,8 @@ Rear channels are mixed more diffusely or attenuated to maintain focus

title="Stereo downmix matrix for the linear surround channel mapping"

align="center">

<artwork align="center"><![CDATA[

- Left output = ( 0.585786 * left + 0.414214 * center )

-Right output = ( 0.414214 * center + 0.585786 * right )

+L output = ( 0.585786 * left + 0.414214 * center )

+R output = ( 0.414214 * center + 0.585786 * right )

]]></artwork>

Exact coefficient values are 1 and 1/sqrt(2), multiplied by

@@ -965,7 +1086,8 @@ The coeffients are in the same order as in <xref target="channel_mapping_1" />,

<t>

The comment header consists of a 64-bit magic signature, followed by data in

the same format as the <xref target="vorbis-comment"/> header used in Ogg

- Vorbis (without the final "framing bit"), Ogg Theora, and Speex.

+ Vorbis, except (like Ogg Theora and Speex) the final "framing bit" specified

+ in the Vorbis spec is not present.

<t><spanx style="strong">Magic Signature</spanx>:

@@ -998,7 +1120,7 @@ It MUST NOT indicate that the vendor string is longer than the rest of the

This is a simple human-readable tag for vendor information, encoded as a UTF-8

string <xref target="RFC3629"/>.

-No terminating null octet is required.

+No terminating null octet is necessary.

This tag is intended to identify the codec encoder and encapsulation

implementations, for tracing differences in technical behavior.

@@ -1041,64 +1163,103 @@ The vendor string length and user comment list length are REQUIRED, and

for these fields, or that do not contain enough data for the corresponding

vendor string or user comments they describe.

Making this check before allocating the associated memory to contain the data

- may help prevent a possible Denial-of-Service (DoS) attack from small comment

+ helps prevent a possible Denial-of-Service (DoS) attack from small comment

headers that claim to contain strings longer than the entire packet or more

user comments than than could possibly fit in the packet.

</t>

<t>

+Immediately following the user comment list, the comment header MAY

+ contain zero-padding or other binary data which is not specified here.

+If the least-significant bit of the first byte of this data is 1, then editors

+ SHOULD preserve the contents of this data when updating the tags, but if this

+ bit is 0, all such data MAY be treated as padding, and truncated or discarded

+ as desired.

+</t>

+<section anchor="comment_format" title="Tag Definitions">

+<t>

The user comment strings follow the NAME=value format described by

- <xref target="vorbis-comment"/> with the same recommended tag names.

-One new comment tag is introduced for Ogg Opus:

+ <xref target="vorbis-comment"/> with the same recommended tag names:

+ ARTIST, TITLE, DATE, ALBUM, and so on.

+</t>

+<t>

+Two new comment tags are introduced here:

+</t>

+ <preamble>An optional gain for track nomalization</preamble>

<artwork align="left"><![CDATA[

R128_TRACK_GAIN=-573

]]></artwork>

-</figure>

-representing the volume shift needed to normalize the track's volume.

+<postamble>

+representing the volume shift needed to normalize the track's volume

+ during isolated playback, in random shuffle, and so on.

The gain is a Q7.8 fixed point number in dB, as in the ID header's 'output

gain' field.

+</postamble>

+</figure>

+<t>

This tag is similar to the REPLAYGAIN_TRACK_GAIN tag in

Vorbis <xref target="replay-gain"/>, except that the normal volume

reference is the <xref target="EBU-R128"/> standard.

</t>

+<figure align="center">

+ <preamble>An optional gain for album nomalization</preamble>

+<artwork align="left"><![CDATA[

+R128_ALBUM_GAIN=111

+]]></artwork>

+<postamble>

+representing the volume shift needed to normalize the overall volume when

+ played as part of a particular collection of tracks.

+The gain is also a Q7.8 fixed point number in dB, as in the ID header's

+ 'output gain' field.

+</postamble>

+</figure>

<t>

-An Ogg Opus file MUST NOT have more than one such tag, and if present its

- value MUST be an integer from -32768 to 32767, inclusive, represented in

- ASCII with no whitespace.

-If present, it MUST correctly represent the R128 normalization gain relative

- to the 'output gain' field specified in the ID header.

-If a player chooses to make use of the R128_TRACK_GAIN tag, it MUST be

- applied <spanx style="emph">in addition</spanx> to the 'output gain' value.

-If an encoder wishes to use R128 normalization, and the output gain is not

- otherwise constrained or specified, the encoder SHOULD write the R128 gain

- into the 'output gain' field and store a tag containing "R128_TRACK_GAIN=0".

-That is, it should assume that by default tools will respect the 'output gain'

- field, and not the comment tag.

+An Ogg Opus stream MUST NOT have more than one of each tag, and if present

+ their values MUST be an integer from -32768 to 32767, inclusive,

+ represented in ASCII as a base 10 number with no whitespace.

+A leading '+' or '-' character is valid.

+Leading zeros are also permitted, but the value MUST be represented by

+ no more than 6 characters.

+Other non-digit characters MUST NOT be present.

+</t>

+<t>

+If present, R128_TRACK_GAIN and R128_ALBUM_GAIN MUST correctly represent

+ the R128 normalization gain relative to the 'output gain' field specified

+ in the ID header.

+If a player chooses to make use of the R128_TRACK_GAIN tag or the

+ R128_ALBUM_GAIN tag, it MUST apply those gains

+ <spanx style="emph">in addition</spanx> to the 'output gain' value.

If a tool modifies the ID header's 'output gain' field, it MUST also update or

- remove the R128_TRACK_GAIN comment tag.

+ remove the R128_TRACK_GAIN and R128_ALBUM_GAIN comment tags if present.

+An encoder SHOULD assume that by default tools will respect the 'output gain'

+ field, and not the comment tag.

</t>

<t>

To avoid confusion with multiple normalization schemes, an Opus comment header

SHOULD NOT contain any of the REPLAYGAIN_TRACK_GAIN, REPLAYGAIN_TRACK_PEAK,

REPLAYGAIN_ALBUM_GAIN, or REPLAYGAIN_ALBUM_PEAK tags.

+<xref target="EBU-R128"/> normalization is preferred to the earlier

+ REPLAYGAIN schemes because of its clear definition and adoption by industry.

+Peak normalizations are difficult to calculate reliably for lossy codecs

+ because of variation in excursion heights due to decoder differences.

+In the authors' investigations they were not applied consistently or broadly

+ enough to merit inclusion here.

</t>

-<t>

-There is no Opus comment tag corresponding to REPLAYGAIN_ALBUM_GAIN.

-That information should instead be stored in the ID header's 'output gain'

- field.

-</t>

-</section>

+</section>

+</section>

-</section>

+</section>

<t>

-Technically valid Opus packets can be arbitrarily large due to the padding

+Technically, valid Opus packets can be arbitrarily large due to the padding

format, although the amount of non-padding data they can contain is bounded.

These packets might be spread over a similarly enormous number of Ogg pages.

-Encoders SHOULD use no more padding than required to make a variable bitrate

- (VBR) stream constant bitrate (CBR).

+Encoders SHOULD use no more padding than is necessary to make a variable

+ bitrate (VBR) stream constant bitrate (CBR).

Decoders SHOULD avoid attempting to allocate excessive amounts of memory when

presented with a very large packet.

The presence of an extremely large packet in the stream could indicate a

@@ -1122,11 +1283,11 @@ Even in such a packet, most of the data will be zeros as 2.5 ms frames

The largest packet consisting of entirely useful data is

(15,326*N - 2) octets, or about 15 kB per stream.

This corresponds to 120 ms of audio encoded as 10 ms frames in either

- LP or Hybrid mode, but at a data rate of over 1 Mbps, which makes little

+ SILK or Hybrid mode, but at a data rate of over 1 Mbps, which makes little

sense for the quality achieved.

A more reasonable limit is (7,664*N - 2) octets, or about 7.5 kB

per stream.

-This corresponds to 120 ms of audio encoded as 20 ms stereo MDCT-mode

+This corresponds to 120 ms of audio encoded as 20 ms stereo CELT mode

frames, with a total bitrate just under 511 kbps (not counting the Ogg

encapsulation overhead).

With N=8, the maximum number of channels currently defined by mapping

@@ -1141,7 +1302,7 @@ An implementation could reasonably choose any of these numbers for its internal

<t>

-When encoding Opus files, Ogg encoders should take into account the

+When encoding Opus streams, Ogg muxers SHOULD take into account the

algorithmic delay of the Opus encoder.

</t>

@@ -1150,18 +1311,19 @@ In encoders derived from the reference implementation, the number of

samples can be queried with:

</preamble>

<artwork align="center"><![CDATA[

- opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD, &samples_delay);

+ opus_encoder_ctl(encoder_state, OPUS_GET_LOOKAHEAD(&delay_samples));

]]></artwork>

</figure>

<t>

To achieve good quality in the very first samples of a stream, the Ogg encoder

- MAY use LPC extrapolation to generate at least 120 extra samples

- (extra_samples) at the beginning to avoid the Opus encoder having to encode

- a discontinuous signal.

-For an input file containing length samples, the Ogg encoder SHOULD set the

- preskip header flag to samples_delay+extra_samples, encode at least

- length+samples_delay+extra_samples samples, and set the granulepos of the last

- page to length+samples_delay+extra_samples.

+ MAY use linear predictive coding (LPC) extrapolation

+ <xref target="linear-prediction"/> to generate at least 120 extra samples at

+ the beginning to avoid the Opus encoder having to encode a discontinuous

+ signal.

+For an input file containing 'length' samples, the Ogg encoder SHOULD set the

+ pre-skip header value to delay_samples+extra_samples, encode at least

+ length+delay_samples+extra_samples samples, and set the granulepos of the last

+ page to length+delay_samples+extra_samples.

This ensures that the encoded file has the same duration as the original, with

no time offset. The best way to pad the end of the stream is to also use LPC

extrapolation, but zero-padding is also acceptable.

@@ -1170,7 +1332,7 @@ This ensures that the encoded file has the same duration as the original, with

<t>

The first step in LPC extrapolation is to compute linear prediction

- coefficients.

+ coefficients. <xref target="lpc-sample"/>

When extending the end of the signal, order-N (typically with N ranging from 8

to 40) LPC analysis is performed on a window near the end of the signal.

The last N samples are used as memory to an infinite impulse response (IIR)

@@ -1205,7 +1367,7 @@ When extending the beginning of the signal, it is best to apply a "fade in" to

<t>

In some applications, such as Internet radio, it is desirable to cut a long

- streams into smaller chains, e.g. so the comment header can be updated.

+ stream into smaller chains, e.g. so the comment header can be updated.

This can be done simply by separating the input streams into segments and

encoding each segment independently.

The drawback of this approach is that it creates a small discontinuity

@@ -1219,12 +1381,26 @@ De-emphasis is allowed.</t>

frame.</t>

<t>Begin the second segment with a copy of the last frame of the first

segment.</t>

-<t>Set the preskip flag of the second stream in such a way as to properly

+<t>Set the pre-skip value of the second stream in such a way as to properly

join the two streams.</t>

<t>Continue the encoding process normally from there, without any reset to

the encoder.</t>

</list>

</t>

+<figure align="center">

+<preamble>

+In encoders derived from the reference implementation, inter-frame prediction

+ can be turned off by calling:

+</preamble>

+<artwork align="center"><![CDATA[

+ opus_encoder_ctl(encoder_state, OPUS_SET_PREDICTION_DISABLED(1));

+]]></artwork>

+<postamble>

+For best results, this implementation requires that prediction be explicitly

+ enabled again before resuming normal encoding, even after a reset.

+</postamble>

+</figure>

</section>

@@ -1237,7 +1413,7 @@ A brief summary of major implementations of this draft is available

</t>

<t>

[Note to RFC Editor: please remove this entire section before

- final publication per <xref target="draft-sheffer-running-code"/>.]

+ final publication per <xref target="RFC6982"/>.]

</t>

</section>

@@ -1248,16 +1424,16 @@ Implementations of the Opus codec need to take appropriate security

This is just as much a problem for the container as it is for the codec itself.

It is extremely important for the decoder to be robust against malicious

payloads.

-Malicious payloads must not cause the decoder to overrun its allocated memory

+Malicious payloads MUST NOT cause the decoder to overrun its allocated memory

or to take an excessive amount of resources to decode.

Although problems in encoders are typically rarer, the same applies to the

encoder.

-Malicious audio streams must not cause the encoder to misbehave because this

+Malicious audio streams MUST NOT cause the encoder to misbehave because this

would allow an attacker to attack transcoding gateways.

</t>

<t>

-Like most other container formats, Ogg Opus files should not be used with

+Like most other container formats, Ogg Opus streams SHOULD NOT be used with

insecure ciphers or cipher modes that are vulnerable to known-plaintext

attacks.

Elements such as the Ogg page capture pattern and the magic signatures in the

@@ -1336,16 +1512,18 @@ The authors agree to grant third parties the irrevocable right to copy, use,

&rfc6381;

&rfc6716;

-<reference anchor="EBU-R128" target="http://tech.ebu.ch/loudness">

+<reference anchor="EBU-R128" target="https://tech.ebu.ch/loudness">

<front>

-<title>"Loudness Recommendation EBU R128</title>

-<author fullname="EBU Technical Committee"/>

-<date month="August" year="2011"/>

+ <title>Loudness Recommendation EBU R128</title>

+ <author>

+ <organization>EBU Technical Committee</organization>

+ </author>

+ <date month="August" year="2011"/>

</front>

</reference>

<reference anchor="vorbis-comment"

- target="http://www.xiph.org/vorbis/doc/v-comment.html">

+ target="https://www.xiph.org/vorbis/doc/v-comment.html">

<front>

<title>Ogg Vorbis I Format Specification: Comment Field and Header

Specification</title>

@@ -1361,16 +1539,7 @@ The authors agree to grant third parties the irrevocable right to copy, use,

&rfc4732;

-<reference anchor="draft-sheffer-running-code"

- target="https://tools.ietf.org/html/draft-sheffer-running-code-05#section-2">

- <front>

- <title>Improving "Rough Consensus" with Running Code</title>

- <author initials="Y." surname="Sheffer" fullname="Yaron Sheffer"/>

- <author initials="A." surname="Farrel" fullname="Adrian Farrel"/>

- <date month="May" year="2013"/>

- </front>

-</reference>

+ &rfc6982;

<reference anchor="flac"

target="https://xiph.org/flac/format.html">

@@ -1382,16 +1551,41 @@ The authors agree to grant third parties the irrevocable right to copy, use,

</reference>

<reference anchor="hanning"

- target="http://en.wikipedia.org/wiki/Hamming_function#Hann_.28Hanning.29_window">

+ target="https://en.wikipedia.org/wiki/Hamming_function#Hann_.28Hanning.29_window">

<front>

- <title>"Hann window</title>

- <author fullname="Wikipedia"/>

+ <title>Hann window</title>

+ <author>

+ <organization>Wikipedia</organization>

+ </author>

</front>

</reference>

+<reference anchor="linear-prediction"

+ target="https://en.wikipedia.org/wiki/Linear_predictive_coding">

+ <front>

+ <title>Linear Predictive Coding</title>

+ <author>

+ <organization>Wikipedia</organization>

+ </author>

+ <date month="January" year="2014"/>

+ </front>

+</reference>

+<reference anchor="lpc-sample"

+ target="https://svn.xiph.org/trunk/vorbis/lib/lpc.c">

+<front>

+ <title>Autocorrelation LPC coeff generation algorithm

+ (Vorbis source code)</title>

+<author initials="J." surname="Degener" fullname="Jutta Degener"/>

+<author initials="C." surname="Bormann" fullname="Carsten Bormann"/>

+<date month="November" year="1994"/>

+</front>

+</reference>

<reference anchor="replay-gain"

- target="http://wiki.xiph.org/VorbisComment#Replay_Gain">

+ target="https://wiki.xiph.org/VorbisComment#Replay_Gain">

<front>

<title>VorbisComment: Replay Gain</title>

@@ -1401,7 +1595,7 @@ The authors agree to grant third parties the irrevocable right to copy, use,

</reference>

<reference anchor="seeking"

- target="http://wiki.xiph.org/Seeking">

+ target="https://wiki.xiph.org/Seeking">

<front>

<title>Granulepos Encoding and How Seeking Really Works</title>

@@ -1412,7 +1606,7 @@ The authors agree to grant third parties the irrevocable right to copy, use,

</reference>

<reference anchor="vorbis-mapping"

- target="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">

+ target="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">

<front>

<title>The Vorbis I Specification, Section 4.3.9 Output Channel Order</title>

<author initials="C." surname="Montgomery"

@@ -1422,7 +1616,7 @@ The authors agree to grant third parties the irrevocable right to copy, use,

</reference>

<reference anchor="vorbis-trim"

- target="http://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-130000A.2">

+ target="https://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-130000A.2">

<front>

<title>The Vorbis I Specification, Appendix A: Embedding Vorbis

into an Ogg stream</title>

@@ -1436,7 +1630,9 @@ The authors agree to grant third parties the irrevocable right to copy, use,

target="http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx">

<front>

<title>Multiple Channel Audio Data and WAVE Files</title>

- <author fullname="Microsoft Corporation"/>

+ <author>

+ <organization>Microsoft Corporation</organization>

+ </author>

</front>

</reference>

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