Vorbis I Specification: February 3, 2012

Vorbis I specification
Xiph.Org Foundation
February 3, 2012
Contents
1. Introduction and Description
1.1. Overview . . . . . . . . . . . . . . . . . . .
1.1.1. Application . . . . . . . . . . . . .
1.1.2. Classification . . . . . . . . . . . .
1.1.3. Assumptions . . . . . . . . . . . . .
1.1.4. Codec Setup and Probability Model
1.1.5. Format Specification . . . . . . . .
1.1.6. Hardware Profile . . . . . . . . . .
1.2. Decoder Configuration . . . . . . . . . . .
1.2.1. Global Config . . . . . . . . . . . .
1.2.2. Mode . . . . . . . . . . . . . . . . .
1.2.3. Mapping . . . . . . . . . . . . . . .
1.2.4. Floor . . . . . . . . . . . . . . . . .
1.2.5. Residue . . . . . . . . . . . . . . .
1.2.6. Codebooks . . . . . . . . . . . . . .
1.3. High-level Decode Process . . . . . . . . .
1.3.1. Decode Setup . . . . . . . . . . . .
1.3.2. Decode Procedure . . . . . . . . . .
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2. Bitpacking Convention
2.1. Overview . . . . . . . . . . . . . . . . .
2.1.1. octets, bytes and words . . . . .
2.1.2. bit order . . . . . . . . . . . . .
2.1.3. byte order . . . . . . . . . . . .
2.1.4. coding bits into byte sequences
2.1.5. signedness . . . . . . . . . . . .
2.1.6. coding example . . . . . . . . .
2.1.7. decoding example . . . . . . . .
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2.1.8. end-of-packet alignment . . . . . . . . . . . . . . . . . . . . . . . .

2.1.9. reading zero bits . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3. Probability Model and Codebooks
3.1. Overview . . . . . . . . . . . . .
3.1.1. Bitwise operation . . . .
3.2. Packed codebook format . . . .
3.2.1. codebook decode . . . .
3.3. Use of the codebook abstraction
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4. Codec Setup and Packet Decode

4.1. Overview . . . . . . . . . . . . . . . . . . . . .
4.2. Header decode and decode setup . . . . . . . .
4.2.1. Common header decode . . . . . . . .
4.2.2. Identification header . . . . . . . . . .
4.2.3. Comment header . . . . . . . . . . . .
4.2.4. Setup header . . . . . . . . . . . . . .
4.3. Audio packet decode and synthesis . . . . . .
4.3.1. packet type, mode and window decode
4.3.2. floor curve decode . . . . . . . . . . . .
4.3.3. nonzero vector propagate . . . . . . . .
4.3.4. residue decode . . . . . . . . . . . . . .
4.3.5. inverse coupling . . . . . . . . . . . . .
4.3.6. dot product . . . . . . . . . . . . . . .
4.3.7. inverse MDCT . . . . . . . . . . . . .
4.3.8. overlap add . . . . . . . . . . . . . . .
4.3.9. output channel order . . . . . . . . . .
5. comment field and header specification
5.1. Overview . . . . . . . . . . . . . . . .
5.2. Comment encoding . . . . . . . . . .
5.2.1. Structure . . . . . . . . . . .
5.2.2. Content vector format . . . .
5.2.3. Encoding . . . . . . . . . . .
6. Floor type 0 setup and decode
6.1. Overview . . . . . . . . . . .
6.2. Floor 0 format . . . . . . . .
6.2.1. header decode . . . .
6.2.2. packet decode . . . .
6.2.3. curve computation .
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7.1. Overview . . . . . . . . . . .
7.2. Floor 1 format . . . . . . . .
7.2.1. model . . . . . . . .
7.2.2. header decode . . . .
7.2.3. packet decode . . . .
7.2.4. curve computation .
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8. Residue setup and decode

8.1. Overview . . . . . . . . .
8.2. Residue format . . . . .
8.3. residue 0 . . . . . . . . .
8.4. residue 1 . . . . . . . . .
8.5. residue 2 . . . . . . . . .
8.6. Residue decode . . . . .
8.6.1. header decode . .
8.6.2. packet decode . .
8.6.3. format 0 specifics
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10.Tables
10.1. floor1 inverse dB table . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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A. Embedding Vorbis into an Ogg stream

A.1. Overview . . . . . . . . . . . . . . .
A.1.1. Restrictions . . . . . . . . .
A.1.2. MIME type . . . . . . . . .
A.2. Encapsulation . . . . . . . . . . . .
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9. Helper equations
9.1. Overview . . . . . . . .
9.2. Functions . . . . . . .
9.2.1. ilog . . . . . . .
9.2.2. float32 unpack
9.2.3. lookup1 values
9.2.4. low neighbor . .
9.2.5. high neighbor .
9.2.6. render point . .
9.2.7. render line . . .
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B. Vorbis encapsulation in RTP
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71
1. Introduction and Description

1.1. Overview
This document provides a high level description of the Vorbis codecs construction. A
bit-by-bit specification appears beginning in Section 4, Codec Setup and Packet Decode.
The later sections assume a high-level understanding of the Vorbis decode process, which
is provided here.
1.1.1. Application
Vorbis is a general purpose perceptual audio CODEC intended to allow maximum encoder flexibility, thus allowing it to scale competitively over an exceptionally wide range
of bitrates. At the high quality/bitrate end of the scale (CD or DAT rate stereo, 16/24
bits) it is in the same league as MPEG-2 and MPC. Similarly, the 1.0 encoder can encode high-quality CD and DAT rate stereo at below 48kbps without resampling to a lower
rate. Vorbis is also intended for lower and higher sample rates (from 8kHz telephony to
192kHz digital masters) and a range of channel representations (monaural, polyphonic,
stereo, quadraphonic, 5.1, ambisonic, or up to 255 discrete channels).
1.1.2. Classification
Vorbis I is a forward-adaptive monolithic transform CODEC based on the Modified Discrete
Cosine Transform. The codec is structured to allow addition of a hybrid wavelet filterbank
in Vorbis II to offer better transient response and reproduction using a transform better
suited to localized time events.
1.1.3. Assumptions
The Vorbis CODEC design assumes a complex, psychoacoustically-aware encoder and simple, low-complexity decoder. Vorbis decode is computationally simpler than mp3, although
it does require more working memory as Vorbis has no static probability model; the vector
codebooks used in the first stage of decoding from the bitstream are packed in their entirety into the Vorbis bitstream headers. In packed form, these codebooks occupy only a
few kilobytes; the extent to which they are pre-decoded into a cache is the dominant factor
in decoder memory usage.
Vorbis provides none of its own framing, synchronization or protection against errors; it is
solely a method of accepting input audio, dividing it into individual frames and compressing
these frames into raw, unformatted packets. The decoder then accepts these raw packets
in sequence, decodes them, synthesizes audio frames from them, and reassembles the frames
into a facsimile of the original audio stream. Vorbis is a free-form variable bit rate (VBR)
codec and packets have no minimum size, maximum size, or fixed/expected size. Packets
are designed that they may be truncated (or padded) and remain decodable; this is not to
be considered an error condition and is used extensively in bitrate management in peeling.
Both the transport mechanism and decoder must allow that a packet may be any size, or
end before or after packet decode expects.
Vorbis packets are thus intended to be used with a transport mechanism that provides
free-form framing, sync, positioning and error correction in accordance with these design
assumptions, such as Ogg (for file transport) or RTP (for network multicast). For purposes
of a few examples in this document, we will assume that Vorbis is to be embedded in an Ogg
stream specifically, although this is by no means a requirement or fundamental assumption
in the Vorbis design.
The specification for embedding Vorbis into an Ogg transport stream is in Appendix A,
Embedding Vorbis into an Ogg stream.
1.1.4. Codec Setup and Probability Model
Vorbis heritage is as a research CODEC and its current design reflects a desire to allow
multiple decades of continuous encoder improvement before running out of room within
the codec specification. For these reasons, configurable aspects of codec setup intentionally
lean toward the extreme of forward adaptive.
The single most controversial design decision in Vorbis (and the most unusual for a Vorbis
developer to keep in mind) is that the entire probability model of the codec, the Huffman
and VQ codebooks, is packed into the bitstream header along with extensive CODEC
setup parameters (often several hundred fields). This makes it impossible, as it would
be with MPEG audio layers, to embed a simple frame type flag in each audio packet, or
begin decode at any frame in the stream without having previously fetched the codec setup
header.
Note: Vorbis can initiate decode at any arbitrary packet within a bitstream so long as
the codec has been initialized/setup with the setup headers.
Thus, Vorbis headers are both required for decode to begin and relatively large as bitstream
headers go. The header size is unbounded, although for streaming a rule-of-thumb of 4kB
or less is recommended (and Xiph.Orgs Vorbis encoder follows this suggestion).
Our own design work indicates the primary liability of the required header is in mindshare;
it is an unusual design and thus causes some amount of complaint among engineers as
this runs against current design trends (and also points out limitations in some existing
software/interface designs, such as Windows ACM codec framework). However, we find

that it does not fundamentally limit Vorbis suitable application space.
1.1.5. Format Specification
The Vorbis format is well-defined by its decode specification; any encoder that produces
packets that are correctly decoded by the reference Vorbis decoder described below may be
considered a proper Vorbis encoder. A decoder must faithfully and completely implement
the specification defined below (except where noted) to be considered a proper Vorbis
decoder.
1.1.6. Hardware Profile
Although Vorbis decode is computationally simple, it may still run into specific limitations
of an embedded design. For this reason, embedded designs are allowed to deviate in limited
ways from the full decode specification yet still be certified compliant. These optional
omissions are labelled in the spec where relevant.
1.2. Decoder Configuration

Decoder setup consists of configuration of multiple, self-contained component abstractions
that perform specific functions in the decode pipeline. Each different component instance
of a specific type is semantically interchangeable; decoder configuration consists both of
internal component configuration, as well as arrangement of specific instances into a decode
pipeline. Componentry arrangement is roughly as follows:
Figure 1: decoder pipeline configuration
1.2.1. Global Config

Global codec configuration consists of a few audio related fields (sample rate, channels),
Vorbis version (always 0 in Vorbis I), bitrate hints, and the lists of component instances.
All other configuration is in the context of specific components.
1.2.2. Mode
Each Vorbis frame is coded according to a master mode. A bitstream may use one or
many modes.
The mode mechanism is used to encode a frame according to one of multiple possible
methods with the intention of choosing a method best suited to that frame. Different
modes are, e.g. how frame size is changed from frame to frame. The mode number of
a frame serves as a top level configuration switch for all other specific aspects of frame
decode.
A mode configuration consists of a frame size setting, window type (always 0, the Vorbis
window, in Vorbis I), transform type (always type 0, the MDCT, in Vorbis I) and a mapping
number. The mapping number specifies which mapping configuration instance to use for
low-level packet decode and synthesis.
1.2.3. Mapping
A mapping contains a channel coupling description and a list of submaps that bundle sets
of channel vectors together for grouped encoding and decoding. These submaps are not
references to external components; the submap list is internal and specific to a mapping.
A submap is a configuration/grouping that applies to a subset of floor and residue vectors
within a mapping. The submap functions as a last layer of indirection such that specific
special floor or residue settings can be applied not only to all the vectors in a given mode,
but also specific vectors in a specific mode. Each submap specifies the proper floor and
residue instance number to use for decoding that submaps spectral floor and spectral
residue vectors.
As an example:
Assume a Vorbis stream that contains six channels in the standard 5.1 format. The sixth
channel, as is normal in 5.1, is bass only. Therefore it would be wasteful to encode a
full-spectrum version of it as with the other channels. The submapping mechanism can
be used to apply a full range floor and residue encoding to channels 0 through 4, and a
bass-only representation to the bass channel, thus saving space. In this example, channels
0-4 belong to submap 0 (which indicates use of a full-range floor) and channel 5 belongs
to submap 1, which uses a bass-only representation.
1.2.4. Floor
Vorbis encodes a spectral floor vector for each PCM channel. This vector is a lowresolution representation of the audio spectrum for the given channel in the current frame,
generally used akin to a whitening filter. It is named a floor because the Xiph.Org
reference encoder has historically used it as a unit-baseline for spectral resolution.
A floor encoding may be of two types. Floor 0 uses a packed LSP representation on a
dB amplitude scale and Bark frequency scale. Floor 1 represents the curve as a piecewise
linear interpolated representation on a dB amplitude scale and linear frequency scale. The
two floors are semantically interchangeable in encoding/decoding. However, floor type 1
provides more stable inter-frame behavior, and so is the preferred choice in all coupledstereo and high bitrate modes. Floor 1 is also considerably less expensive to decode than
floor 0.
Floor 0 is not to be considered deprecated, but it is of limited modern use. No known
Vorbis encoder past Xiph.Orgs own beta 4 makes use of floor 0.
The values coded/decoded by a floor are both compactly formatted and make use of entropy coding to save space. For this reason, a floor configuration generally refers to multiple codebooks in the codebook component list. Entropy coding is thus provided as an
abstraction, and each floor instance may choose from any and all available codebooks when
coding/decoding.
1.2.5. Residue
The spectral residue is the fine structure of the audio spectrum once the floor curve has been
subtracted out. In simplest terms, it is coded in the bitstream using cascaded (multi-pass)
vector quantization according to one of three specific packing/coding algorithms numbered
0 through 2. The packing algorithm details are configured by residue instance. As with
the floor components, the final VQ/entropy encoding is provided by external codebook
instances and each residue instance may choose from any and all available codebooks.
1.2.6. Codebooks
Codebooks are a self-contained abstraction that perform entropy decoding and, optionally,
use the entropy-decoded integer value as an offset into an index of output value vectors,
returning the indicated vector of values.
The entropy coding in a Vorbis I codebook is provided by a standard Huffman binary tree
representation. This tree is tightly packed using one of several methods, depending on
whether codeword lengths are ordered or unordered, or the tree is sparse.
The codebook vector index is similarly packed according to index characteristic. Most
commonly, the vector index is encoded as a single list of values of possible values that are
then permuted into a list of n-dimensional rows (lattice VQ).
1.3. High-level Decode Process

1.3.1. Decode Setup
Before decoding can begin, a decoder must initialize using the bitstream headers matching
the stream to be decoded. Vorbis uses three header packets; all are required, in-order, by
this specification. Once set up, decode may begin at any audio packet belonging to the
Vorbis stream. In Vorbis I, all packets after the three initial headers are audio packets.
The header packets are, in order, the identification header, the comments header, and the
setup header.
Identification Header The identification header identifies the bitstream as Vorbis, Vorbis
version, and the simple audio characteristics of the stream such as sample rate and number
of channels.
Comment Header The comment header includes user text comments (tags) and a
vendor string for the application/library that produced the bitstream. The encoding and
proper use of the comment header is described in Section 5, comment field and header
specification.
Setup Header The setup header includes extensive CODEC setup information as well
as the complete VQ and Huffman codebooks needed for decode.
1.3.2. Decode Procedure
The decoding and synthesis procedure for all audio packets is fundamentally the same.
1. decode packet type flag
2. decode mode number
3. decode window shape (long windows only)
4. decode floor
5. decode residue into residue vectors
6. inverse channel coupling of residue vectors

7. generate floor curve from decoded floor data
8. compute dot product of floor and residue, producing audio spectrum vector
9. inverse monolithic transform of audio spectrum vector, always an MDCT in Vorbis I
10. overlap/add left-hand output of transform with right-hand output of previous frame
11. store right hand-data from transform of current frame for future lapping
12. if not first frame, return results of overlap/add as audio result of current frame
Note that clever rearrangement of the synthesis arithmetic is possible; as an example, one
can take advantage of symmetries in the MDCT to store the right-hand transform data of a
partial MDCT for a 50% inter-frame buffer space savings, and then complete the transform
later before overlap/add with the next frame. This optimization produces entirely equivalent output and is naturally perfectly legal. The decoder must be entirely mathematically
equivalent to the specification, it need not be a literal semantic implementation.
Packet type decode Vorbis I uses four packet types. The first three packet types mark
each of the three Vorbis headers described above. The fourth packet type marks an audio
packet. All other packet types are reserved; packets marked with a reserved type should
be ignored.
Following the three header packets, all packets in a Vorbis I stream are audio. The first
step of audio packet decode is to read and verify the packet type; a non-audio packet when
audio is expected indicates stream corruption or a non-compliant stream. The decoder must
ignore the packet and not attempt decoding it to audio.
Mode decode Vorbis allows an encoder to set up multiple, numbered packet modes, as
described earlier, all of which may be used in a given Vorbis stream. The mode is encoded
as an integer used as a direct offset into the mode instance index.
Window shape decode (long windows only) Vorbis frames may be one of two PCM
sample sizes specified during codec setup. In Vorbis I, legal frame sizes are powers of two
from 64 to 8192 samples. Aside from coupling, Vorbis handles channels as independent
vectors and these frame sizes are in samples per channel.
Vorbis uses an overlapping transform, namely the MDCT, to blend one frame into the
next, avoiding most inter-frame block boundary artifacts. The MDCT output of one frame
is windowed according to MDCT requirements, overlapped 50% with the output of the
previous frame and added. The window shape assures seamless reconstruction.
This is easy to visualize in the case of equal sized-windows:
10
Figure 2: overlap of two equal-sized windows

And slightly more complex in the case of overlapping unequal sized windows:
Figure 3: overlap of a long and a short window

In the unequal-sized window case, the window shape of the long window must be modified
for seamless lapping as above. It is possible to correctly infer window shape to be applied
to the current window from knowing the sizes of the current, previous and next window.
It is legal for a decoder to use this method. However, in the case of a long window (short
windows require no modification), Vorbis also codes two flag bits to specify pre- and postwindow shape. Although not strictly necessary for function, this minor redundancy allows
a packet to be fully decoded to the point of lapping entirely independently of any other
packet, allowing easier abstraction of decode layers as well as allowing a greater level of
easy parallelism in encode and decode.
A description of valid window functions for use with an inverse MDCT can be found in [1].
Vorbis windows all use the slope function
y = sin(.5 sin2 ((x + .5)/n )).
floor decode Each floor is encoded/decoded in channel order, however each floor belongs
to a submap that specifies which floor configuration to use. All floors are decoded before
residue decode begins.
11
residue decode Although the number of residue vectors equals the number of channels,
channel coupling may mean that the raw residue vectors extracted during decode do not
map directly to specific channels. When channel coupling is in use, some vectors will
correspond to coupled magnitude or angle. The coupling relationships are described in the
codec setup and may differ from frame to frame, due to different mode numbers.
Vorbis codes residue vectors in groups by submap; the coding is done in submap order
from submap 0 through n-1. This differs from floors which are coded using a configuration
provided by submap number, but are coded individually in channel order.
inverse channel coupling A detailed discussion of stereo in the Vorbis codec can be found
in the document Stereo Channel Coupling in the Vorbis CODEC. Vorbis is not limited to
only stereo coupling, but the stereo document also gives a good overview of the generic
coupling mechanism.
Vorbis coupling applies to pairs of residue vectors at a time; decoupling is done in-place
a pair at a time in the order and using the vectors specified in the current mapping
configuration. The decoupling operation is the same for all pairs, converting square polar
representation (where one vector is magnitude and the second angle) back to Cartesian
representation.
After decoupling, in order, each pair of vectors on the coupling list, the resulting residue
vectors represent the fine spectral detail of each output channel.
generate floor curve The decoder may choose to generate the floor curve at any appropriate time. It is reasonable to generate the output curve when the floor data is decoded from
the raw packet, or it can be generated after inverse coupling and applied to the spectral
residue directly, combining generation and the dot product into one step and eliminating
some working space.
Both floor 0 and floor 1 generate a linear-range, linear-domain output vector to be multiplied (dot product) by the linear-range, linear-domain spectral residue.
compute floor/residue dot product This step is straightforward; for each output channel, the decoder multiplies the floor curve and residue vectors element by element, producing the finished audio spectrum of each channel.
One point is worth mentioning about this dot product; a common mistake in a fixed point
implementation might be to assume that a 32 bit fixed-point representation for floor and
residue and direct multiplication of the vectors is sufficient for acceptable spectral depth in
all cases because it happens to mostly work with the current Xiph.Org reference encoder.
However, floor vector values can span 140dB (24 bits unsigned), and the audio spectrum
vector should represent a minimum of 120dB (21 bits with sign), even when output is to
12
a 16 bit PCM device. For the residue vector to represent full scale if the floor is nailed to
140dB, it must be able to span 0 to +140dB. For the residue vector to reach full scale if
the floor is nailed at 0dB, it must be able to represent 140dB to +0dB. Thus, in order to
handle full range dynamics, a residue vector may span 140dB to +140dB entirely within
spec. A 280dB range is approximately 48 bits with sign; thus the residue vector must be
able to represent a 48 bit range and the dot product must be able to handle an effective 48
bit times 24 bit multiplication. This range may be achieved using large (64 bit or larger)
integers, or implementing a movable binary point representation.
inverse monolithic transform (MDCT) The audio spectrum is converted back into time
domain PCM audio via an inverse Modified Discrete Cosine Transform (MDCT). A detailed
description of the MDCT is available in [1].
Note that the PCM produced directly from the MDCT is not yet finished audio; it must be
lapped with surrounding frames using an appropriate window (such as the Vorbis window)
before the MDCT can be considered orthogonal.
overlap/add data Windowed MDCT output is overlapped and added with the right hand
data of the previous window such that the 3/4 point of the previous window is aligned with
the 1/4 point of the current window (as illustrated in the window overlap diagram). At
this point, the audio data between the center of the previous frame and the center of the
current frame is now finished and ready to be returned.
cache right hand data The decoder must cache the right hand portion of the current
frame to be lapped with the left hand portion of the next frame.
return finished audio data The overlapped portion produced from overlapping the previous and current frame data is finished data to be returned by the decoder. This data
spans from the center of the previous window to the center of the current window. In the
case of same-sized windows, the amount of data to return is one-half block consisting of
and only of the overlapped portions. When overlapping a short and long window, much of
the returned range is not actually overlap. This does not damage transform orthogonality. Pay attention however to returning the correct data range; the amount of data to be
returned is:
1
window_blocksize(previous_window)/4+window_blocksize(current_window)/4
from the center of the previous window to the center of the current window.
Data is not returned from the first frame; it must be used to prime the decode engine.
The encoder accounts for this priming when calculating PCM offsets; after the first frame,
the proper PCM output offset is 0 (as no data has been returned yet).
13
2. Bitpacking Convention
2.1. Overview
The Vorbis codec uses relatively unstructured raw packets containing arbitrary-width binary integer fields. Logically, these packets are a bitstream in which bits are coded one-byone by the encoder and then read one-by-one in the same monotonically increasing order
by the decoder. Most current binary storage arrangements group bits into a native word
size of eight bits (octets), sixteen bits, thirty-two bits or, less commonly other fixed word
sizes. The Vorbis bitpacking convention specifies the correct mapping of the logical packet
bitstream into an actual representation in fixed-width words.
2.1.1. octets, bytes and words
In most contemporary architectures, a byte is synonymous with an octet, that is, eight
bits. This has not always been the case; seven, ten, eleven and sixteen bit bytes have
been used. For purposes of the bitpacking convention, a byte implies the native, smallest
integer storage representation offered by a platform. On modern platforms, this is generally assumed to be eight bits (not necessarily because of the processor but because of
the filesystem/memory architecture. Modern filesystems invariably offer bytes as the fundamental atom of storage). A word is an integer size that is a grouped multiple of this
smallest size.
The most ubiquitous architectures today consider a byte to be an octet (eight bits) and
a word to be a group of two, four or eight bytes (16, 32 or 64 bits). Note however that
the Vorbis bitpacking convention is still well defined for any native byte size; Vorbis uses
the native bit-width of a given storage system. This document assumes that a byte is one
octet for purposes of example.
2.1.2. bit order
A byte has a well-defined least significant bit (LSb), which is the only bit set when the
byte is storing the twos complement integer value +1. A bytes most significant bit
(MSb) is at the opposite end of the byte. Bits in a byte are numbered from zero at the
LSb to n (n = 7 in an octet) for the MSb.
2.1.3. byte order
Words are native groupings of multiple bytes. Several byte orderings are possible in a
word; the common ones are 3-2-1-0 (big endian or most significant byte first in which
14
the highest-valued byte comes first), 0-1-2-3 (little endian or least significant byte first
in which the lowest value byte comes first) and less commonly 3-1-2-0 and 0-2-1-3 (mixed
endian).
The Vorbis bitpacking convention specifies storage and bitstream manipulation at the byte,
not word, level, thus host word ordering is of a concern only during optimization when
writing high performance code that operates on a word of storage at a time rather than by
byte. Logically, bytes are always coded and decoded in order from byte zero through byte
n.
2.1.4. coding bits into byte sequences
The Vorbis codec has need to code arbitrary bit-width integers, from zero to 32 bits wide,
into packets. These integer fields are not aligned to the boundaries of the byte representation; the next field is written at the bit position at which the previous field ends.
The encoder logically packs integers by writing the LSb of a binary integer to the logical
bitstream first, followed by next least significant bit, etc, until the requested number of
bits have been coded. When packing the bits into bytes, the encoder begins by placing
the LSb of the integer to be written into the least significant unused bit position of the
destination byte, followed by the next-least significant bit of the source integer and so on
up to the requested number of bits. When all bits of the destination byte have been filled,
encoding continues by zeroing all bits of the next byte and writing the next bit into the
bit position 0 of that byte. Decoding follows the same process as encoding, but by reading
bits from the byte stream and reassembling them into integers.
2.1.5. signedness
The signedness of a specific number resulting from decode is to be interpreted by the
decoder given decode context. That is, the three bit binary pattern b111 can be taken
to represent either seven as an unsigned integer, or -1 as a signed, twos complement
integer. The encoder and decoder are responsible for knowing if fields are to be treated as
signed or unsigned.
2.1.6. coding example
Code the 4 bit integer value 12 [b1100] into an empty bytestream. Bytestream result:
1
2
|
V
3
4
5
6
7 6 5 4 3 2 1 0
byte 0 [0 0 0 0 1 1 0 0]
byte 1 [
]
<-
15
7
8
byte 2 [
byte 3 [
]
]
...
9
10
byte n [
bytestream length == 1 byte
11
Continue by coding the 3 bit integer value -1 [b111]:

|
V
1
2
3
4
5
6
7
8
9
10
7 6 5 4 3 2 1 0
[0 1 1 1 1 1 0 0]
[
]
[
]
[
]
...
byte n [
]
byte
byte
byte
byte
0
1
2
3
<-
bytestream length == 1 byte
Continue by coding the 7 bit integer value 17 [b0010001]:

|
V
1
2
3
4
5
6
7
8
9
10
7 6 5 4 3 2 1 0
[1 1 1 1 1 1 0 0]
[0 0 0 0 1 0 0 0]
[
]
[
]
...
byte n [
]
byte
byte
byte
byte
0
1
2
3
11
<-
bytestream length == 2 bytes

bit cursor == 6
Continue by coding the 13 bit integer value 6969 [b110 11001110 01]:
|
V
1
2
3
4
5
6
7
8
byte
byte
byte
byte
0
1
2
3
7
[1
[0
[1
[0
6
1
1
1
0
9
10
5 4 3
1 1 1
0 0 1
0 0 1
0 0 0
...
2
1
0
1
1
1
0
0
1
1
byte n [
0
0]
0]
0]
0]
]
<bytestream length == 4 bytes
11
2.1.7. decoding example

Reading from the beginning of the bytestream encoded in the above example:
|
V
1
2
3
4
5
6
7
8
byte
byte
byte
byte
0
1
2
3
7
[1
[0
[1
[0
6
1
1
1
0
5
1
0
0
0
4
1
0
0
0
3
1
1
1
0
2
1
0
1
1
1
0
0
1
1
0
0]
0]
0]
0]
<-
bytestream length == 4 bytes
16
We read two, two-bit integer fields, resulting in the returned numbers b00 and b11. Two
things are worth noting here:
Although these four bits were originally written as a single four-bit integer, reading
some other combination of bit-widths from the bitstream is well defined. There are
no artificial alignment boundaries maintained in the bitstream.
The second value is the two-bit-wide integer b11. This value may be interpreted
either as the unsigned value 3, or the signed value -1. Signedness is dependent on
decode context.
2.1.8. end-of-packet alignment
The typical use of bitpacking is to produce many independent byte-aligned packets which
are embedded into a larger byte-aligned container structure, such as an Ogg transport
bitstream. Externally, each bytestream (encoded bitstream) must begin and end on a byte
boundary. Often, the encoded bitstream is not an integer number of bytes, and so there is
unused (uncoded) space in the last byte of a packet.
Unused space in the last byte of a bytestream is always zeroed during the coding process.
Thus, should this unused space be read, it will return binary zeroes.
Attempting to read past the end of an encoded packet results in an end-of-packet condition. End-of-packet is not to be considered an error; it is merely a state indicating that
there is insufficient remaining data to fulfill the desired read size. Vorbis uses truncated
packets as a normal mode of operation, and as such, decoders must handle reading past
the end of a packet as a typical mode of operation. Any further read operations after an
end-of-packet condition shall also return end-of-packet.
2.1.9. reading zero bits
Reading a zero-bit-wide integer returns the value 0 and does not increment the stream
cursor. Reading to the end of the packet (but not past, such that an end-of-packet
condition has not triggered) and then reading a zero bit integer shall succeed, returning 0,
and not trigger an end-of-packet condition. Reading a zero-bit-wide integer after a previous
read sets end-of-packet shall also fail with end-of-packet.
17
3. Probability Model and Codebooks

3.1. Overview
Unlike practically every other mainstream audio codec, Vorbis has no statically configured
probability model, instead packing all entropy decoding configuration, VQ and Huffman,
into the bitstream itself in the third header, the codec setup header. This packed configuration consists of multiple codebooks, each containing a specific Huffman-equivalent
representation for decoding compressed codewords as well as an optional lookup table of
output vector values to which a decoded Huffman value is applied as an offset, generating
the final decoded output corresponding to a given compressed codeword.
3.1.1. Bitwise operation
The codebook mechanism is built on top of the vorbis bitpacker. Both the codebooks themselves and the codewords they decode are unrolled from a packet as a series of arbitrarywidth values read from the stream according to Section 2, Bitpacking Convention.
3.2. Packed codebook format

For purposes of the examples below, we assume that the storage systems native byte
width is eight bits. This is not universally true; see Section 2, Bitpacking Convention
for discussion relating to non-eight-bit bytes.
3.2.1. codebook decode
A codebook begins with a 24 bit sync pattern, 0x564342:
1
2
3
byte 0: [ 0 1 0 0 0 0 1 0 ] (0x42)
byte 1: [ 0 1 0 0 0 0 1 1 ] (0x43)
byte 2: [ 0 1 0 1 0 1 1 0 ] (0x56)
16 bit [codebook_dimensions] and 24 bit [codebook_entries] fields:

1
2
3
byte 3: [ X X X X X X X X ]
byte 4: [ X X X X X X X X ] [codebook_dimensions] (16 bit unsigned)
4
5
6
7
byte 7: [ X X X X X X X X ] [codebook_entries] (24 bit unsigned)
Next is the [ordered] bit flag:
18
1
2
byte 8: [
X ] [ordered] (1 bit)
Each entry, numbering a total of [codebook_entries], is assigned a codeword length.

We now read the list of codeword lengths and store these lengths in the array [codebook_
codeword_lengths]. Decode of lengths is according to whether the [ordered] flag is set
or unset.
If the [ordered] flag is unset, the codeword list is not length ordered and the decoder
needs to read each codeword length one-by-one.
The decoder first reads one additional bit flag, the [sparse] flag. This flag determines whether or not the codebook contains unused entries that are not to be
included in the codeword decode tree:
1
byte 8: [
X 1 ] [sparse] flag (1 bit)
The decoder now performs for each of the [codebook_entries] codebook entries:
1
2
1) if([sparse] is set) {
2) [flag] = read one bit;

3) if([flag] is set) {
4
5
6
4) [length] = read a five bit unsigned integer;

5) codeword length for this entry is [length]+1;
7
8
9
} else {
10
11
6) this entry is unused.
12
mark it as such.
13
14
15
16
} else the sparse flag is not set {
17
7) [length] = read a five bit unsigned integer;

8) the codeword length for this entry is [length]+1;
18
19
20
21
22
If the [ordered] flag is set, the codeword list for this codebook is encoded in ascending length order. Rather than reading a length for every codeword, the encoder
reads the number of codewords per length. That is, beginning at entry zero:
1
2
3
4
5
6
7
8
9
10
11
1) [current_entry] = 0;
2) [current_length] = read a five bit unsigned integer and add 1;
3) [number] = read ilog([codebook_entries] - [current_entry]) bits as an unsigned integer
4) set the entries [current_entry] through [current_entry]+[number]-1, inclusive,
of the [codebook_codeword_lengths] array to [current_length]
5) set [current_entry] to [number] + [current_entry]
6) increment [current_length] by 1
7) if [current_entry] is greater than [codebook_entries] ERROR CONDITION;
the decoder will not be able to read this stream.
8) if [current_entry] is less than [codebook_entries], repeat process starting at 3)
9) done.
19
After all codeword lengths have been decoded, the decoder reads the vector lookup table.
Vorbis I supports three lookup types:
1. No lookup
2. Implicitly populated value mapping (lattice VQ)
3. Explicitly populated value mapping (tessellated or foam VQ)
The lookup table type is read as a four bit unsigned integer:
1
1) [codebook_lookup_type] = read four bits as an unsigned integer
Codebook decode precedes according to [codebook_lookup_type]:

Lookup type zero indicates no lookup to be read. Proceed past lookup decode.
Lookup types one and two are similar, differing only in the number of lookup values to
be read. Lookup type one reads a list of values that are permuted in a set pattern to
build a list of vectors, each vector of order [codebook_dimensions] scalars. Lookup
type two builds the same vector list, but reads each scalar for each vector explicitly,
rather than building vectors from a smaller list of possible scalar values. Lookup
decode proceeds as follows:
1
2
3
4
1)
2)
3)
4)
[codebook_minimum_value] = float32_unpack( read 32 bits as an unsigned integer)

[codebook_delta_value] = float32_unpack( read 32 bits as an unsigned integer)
[codebook_value_bits] = read 4 bits as an unsigned integer and add 1
[codebook_sequence_p] = read 1 bit as a boolean flag
5
6
if ( [codebook_lookup_type] is 1 ) {
5) [codebook_lookup_values] = lookup1_values([codebook_entries], [codebook_dimensions] )
8
9
10
} else {
11
6) [codebook_lookup_values] = [codebook_entries] * [codebook_dimensions]
12
13
14
15
16
17
7) read a total of [codebook_lookup_values] unsigned integers of [codebook_value_bits] each;

store these in order in the array [codebook_multiplicands]
A [codebook_lookup_type] of greater than two is reserved and indicates a stream

that is not decodable by the specification in this document.
An end of packet during any read operation in the above steps is considered an error
condition rendering the stream undecodable.
Huffman decision tree representation The [codebook_codeword_lengths] array and
[codebook_entries] value uniquely define the Huffman decision tree used for entropy
decoding.
Briefly, each used codebook entry (recall that length-unordered codebooks support unused
codeword entries) is assigned, in order, the lowest valued unused binary Huffman codeword
20
possible. Assume the following codeword length list:

1
2
3
4
5
6
7
8
entry
entry
entry
entry
entry
entry
entry
entry
0:
1:
2:
3:
4:
5:
6:
7:
length
length
length
length
length
length
length
length
2
4
4
4
4
2
3
3
Assigning codewords in order (lowest possible value of the appropriate length to highest)
results in the following codeword list:
1
2
3
4
5
6
7
8
entry
entry
entry
entry
entry
entry
entry
entry
0:
1:
2:
3:
4:
5:
6:
7:
length
length
length
length
length
length
length
length
2
4
4
4
4
2
3
3
codeword
codeword
codeword
codeword
codeword
codeword
codeword
codeword
00
0100
0101
0110
0111
10
110
111
Note: Unlike most binary numerical values in this document, we intend the above codewords to be read and used bit by bit from left to right, thus the codeword 001 is the
bit string zero, zero, one. When determining lowest possible value in the assignment
definition above, the leftmost bit is the MSb.
It is clear that the codeword length list represents a Huffman decision tree with the entry
numbers equivalent to the leaves numbered left-to-right:
Figure 4: huffman tree illustration

As we assign codewords in order, we see that each choice constructs a new leaf in the
leftmost possible position.
21
Note that its possible to underspecify or overspecify a Huffman tree via the length list. In
the above example, if codeword seven were eliminated, its clear that the tree is unfinished:
Figure 5: underspecified huffman tree illustration

Similarly, in the original codebook, its clear that the tree is fully populated and a ninth
codeword is impossible. Both underspecified and overspecified trees are an error condition
rendering the stream undecodable. Take special care that a codebook with a single used
entry is handled properly; it consists of a single codework of zero bits and reading a value
out of such a codebook always returns the single used value and sinks zero bits.
Codebook entries marked unused are simply skipped in the assigning process. They have
no codeword and do not appear in the decision tree, thus its impossible for any bit pattern
read from the stream to decode to that entry number.
VQ lookup table vector representation Unpacking the VQ lookup table vectors relies
on the following values:
1
2
3
4
5
6
7
8
the [codebook\_multiplicands] array

[codebook\_minimum\_value]
[codebook\_delta\_value]
[codebook\_sequence\_p]
[codebook\_lookup\_type]
[codebook\_entries]
[codebook\_dimensions]
[codebook\_lookup\_values]
Decoding (unpacking) a specific vector in the vector lookup table proceeds according to
[codebook_lookup_type]. The unpacked vector values are what a codebook would return
during audio packet decode in a VQ context.
22
Vector value decode: Lookup type 1 Lookup type one specifies a lattice VQ lookup
table built algorithmically from a list of scalar values. Calculate (unpack) the final values
of a codebook entry vector from the entries in [codebook_multiplicands] as follows
([value_vector] is the output vector representing the vector of values for entry number
[lookup_offset] in this codebook):
1
2
3
1) [last] = 0;
2) [index_divisor] = 1;
3) iterate [i] over the range 0 ... [codebook_dimensions]-1 (once for each scalar value in the value vector) {
4) [multiplicand_offset] = ( [lookup_offset] divided by [index_divisor] using integer

division ) integer modulo [codebook_lookup_values]
5
6
7
5) vector [value_vector] element [i] =

( [codebook_multiplicands] array element number [multiplicand_offset] ) *
[codebook_delta_value] + [codebook_minimum_value] + [last];
8
9
10
11
6) if ( [codebook_sequence_p] is set ) then set [last] = vector [value_vector] element [i]
12
13
7) [index_divisor] = [index_divisor] * [codebook_lookup_values]
14
15
16
17
18
8) vector calculation completed.
Vector value decode: Lookup type 2 Lookup type two specifies a VQ lookup table
in which each scalar in each vector is explicitly set by the [codebook_multiplicands]
array in a one-to-one mapping. Calculate [unpack] the final values of a codebook entry
vector from the entries in [codebook_multiplicands] as follows ([value_vector] is the
output vector representing the vector of values for entry number [lookup_offset] in this
codebook):
1
2
3
1) [last] = 0;
2) [multiplicand_offset] = [lookup_offset] * [codebook_dimensions]
3) iterate [i] over the range 0 ... [codebook_dimensions]-1 (once for each scalar value in the value vector) {
4) vector [value_vector] element [i] =

( [codebook_multiplicands] array element number [multiplicand_offset] ) *
[codebook_delta_value] + [codebook_minimum_value] + [last];
5
6
7
8
5) if ( [codebook_sequence_p] is set ) then set [last] = vector [value_vector] element [i]
9
10
6) increment [multiplicand_offset]
11
12
13
14
15
7) vector calculation completed.
3.3. Use of the codebook abstraction

The decoder uses the codebook abstraction much as it does the bit-unpacking convention;
a specific codebook reads a codeword from the bitstream, decoding it into an entry number,
and then returns that entry number to the decoder (when used in a scalar entropy coding
23
context), or uses that entry number as an offset into the VQ lookup table, returning a
vector of values (when used in a context desiring a VQ value). Scalar or VQ context is
always explicit; any call to the codebook mechanism requests either a scalar entry number
or a lookup vector.
Note that VQ lookup type zero indicates that there is no lookup table; requesting decode
using a codebook of lookup type 0 in any context expecting a vector return value (even in
a case where a vector of dimension one) is forbidden. If decoder setup or decode requests
such an action, that is an error condition rendering the packet undecodable.
Using a codebook to read from the packet bitstream consists first of reading and decoding
the next codeword in the bitstream. The decoder reads bits until the accumulated bits
match a codeword in the codebook. This process can be though of as logically walking the
Huffman decode tree by reading one bit at a time from the bitstream, and using the bit as
a decision boolean to take the 0 branch (left in the above examples) or the 1 branch (right
in the above examples). Walking the tree finishes when the decode process hits a leaf in
the decision tree; the result is the entry number corresponding to that leaf. Reading past
the end of a packet propagates the end-of-stream condition to the decoder.
When used in a scalar context, the resulting codeword entry is the desired return value.
When used in a VQ context, the codeword entry number is used as an offset into the VQ
lookup table. The value returned to the decoder is the vector of scalars corresponding to
this offset.
24
4. Codec Setup and Packet Decode

4.1. Overview
This document serves as the top-level reference document for the bit-by-bit decode specification of Vorbis I. This document assumes a high-level understanding of the Vorbis decode
process, which is provided in Section 1, Introduction and Description. Section 2, Bitpacking Convention covers reading and writing bit fields from and to bitstream packets.
4.2. Header decode and decode setup

A Vorbis bitstream begins with three header packets. The header packets are, in order,
the identification header, the comments header, and the setup header. All are required for
decode compliance. An end-of-packet condition during decoding the first or third header
packet renders the stream undecodable. End-of-packet decoding the comment header is a
non-fatal error condition.
4.2.1. Common header decode
Each header packet begins with the same header fields.
1
2
1) [packet_type] : 8 bit value

2) 0x76, 0x6f, 0x72, 0x62, 0x69, 0x73: the characters v,o,r,b,i,s as six octets
Decode continues according to packet type; the identification header is type 1, the comment
header type 3 and the setup header type 5 (these types are all odd as a packet with a leading
single bit of 0 is an audio packet). The packets must occur in the order of identification,
comment, setup.
4.2.2. Identification header
The identification header is a short header of only a few fields used to declare the stream
definitively as Vorbis, and provide a few externally relevant pieces of information about
the audio stream. The identification header is coded as follows:
1
2
3
4
5
6
7
8
9
1)
2)
3)
4)
5)
6)
7)
8)
9)
[vorbis_version] = read 32 bits as unsigned integer

[audio_channels] = read 8 bit integer as unsigned
[audio_sample_rate] = read 32 bits as unsigned integer
[bitrate_maximum] = read 32 bits as signed integer
[bitrate_nominal] = read 32 bits as signed integer
[bitrate_minimum] = read 32 bits as signed integer
[blocksize_0] = 2 exponent (read 4 bits as unsigned integer)
[blocksize_1] = 2 exponent (read 4 bits as unsigned integer)
[framing_flag] = read one bit
25
[vorbis_version] is to read 0 in order to be compatible with this document. Both

[audio_channels] and [audio_sample_rate] must read greater than zero. Allowed final
blocksize values are 64, 128, 256, 512, 1024, 2048, 4096 and 8192 in Vorbis I. [blocksize_
0] must be less than or equal to [blocksize_1]. The framing bit must be nonzero. Failure
to meet any of these conditions renders a stream undecodable.
The bitrate fields above are used only as hints. The nominal bitrate field especially may
be considerably off in purely VBR streams. The fields are meaningful only when greater
than zero.
All three fields set to the same value implies a fixed rate, or tightly bounded, nearly
fixed-rate bitstream
Only nominal set implies a VBR or ABR stream that averages the nominal bitrate
Maximum and or minimum set implies a VBR bitstream that obeys the bitrate limits
None set indicates the encoder does not care to speculate.
4.2.3. Comment header
Comment header decode and data specification is covered in Section 5, comment field and
header specification.
4.2.4. Setup header
Vorbis codec setup is configurable to an extreme degree:
Figure 6: decoder pipeline configuration

The setup header contains the bulk of the codec setup information needed for decode. The
setup header contains, in order, the lists of codebook configurations, time-domain transform configurations (placeholders in Vorbis I), floor configurations, residue configurations,
channel mapping configurations and mode configurations. It finishes with a framing bit of
1. Header decode proceeds in the following order:
26
Codebooks
1. [vorbis_codebook_count] = read eight bits as unsigned integer and add one
2. Decode [vorbis_codebook_count] codebooks in order as defined in Section 3, Probability Model and Codebooks. Save each configuration, in order, in an array of
codebook configurations [vorbis_codebook_configurations].
Time domain transforms These hooks are placeholders in Vorbis I. Nevertheless, the
configuration placeholder values must be read to maintain bitstream sync.
1. [vorbis_time_count] = read 6 bits as unsigned integer and add one
2. read [vorbis_time_count] 16 bit values; each value should be zero. If any value is
nonzero, this is an error condition and the stream is undecodable.
Floors Vorbis uses two floor types; header decode is handed to the decode abstraction of
the appropriate type.
1. [vorbis_floor_count] = read 6 bits as unsigned integer and add one
2. For each [i] of [vorbis_floor_count] floor numbers:
a) read the floor type: vector [vorbis_floor_types] element [i] = read 16 bits
as unsigned integer
b) If the floor type is zero, decode the floor configuration as defined in Section 6,
Floor type 0 setup and decode; save this configuration in slot [i] of the floor
configuration array [vorbis_floor_configurations].
c) If the floor type is one, decode the floor configuration as defined in Section 7,
Floor type 1 setup and decode; save this configuration in slot [i] of the floor
configuration array [vorbis_floor_configurations].
d) If the the floor type is greater than one, this stream is undecodable; ERROR
CONDITION
Residues Vorbis uses three residue types; header decode of each type is identical.
1. [vorbis_residue_count] = read 6 bits as unsigned integer and add one
2. For each of [vorbis_residue_count] residue numbers:
a) read the residue type; vector [vorbis_residue_types] element [i] = read 16
bits as unsigned integer
27
b) If the residue type is zero, one or two, decode the residue configuration as defined
in Section 8, Residue setup and decode; save this configuration in slot [i] of
the residue configuration array [vorbis_residue_configurations].
c) If the the residue type is greater than two, this stream is undecodable; ERROR
CONDITION
Mappings Mappings are used to set up specific pipelines for encoding multichannel audio
with varying channel mapping applications. Vorbis I uses a single mapping type (0), with
implicit PCM channel mappings.
1. [vorbis_mapping_count] = read 6 bits as unsigned integer and add one
2. For each [i] of [vorbis_mapping_count] mapping numbers:
a) read the mapping type: 16 bits as unsigned integer. Theres no reason to save
the mapping type in Vorbis I.
b) If the mapping type is nonzero, the stream is undecodable
c) If the mapping type is zero:
i. read 1 bit as a boolean flag
A. if set, [vorbis_mapping_submaps] = read 4 bits as unsigned integer
and add one
B. if unset, [vorbis_mapping_submaps] = 1
ii. read 1 bit as a boolean flag
A. if set, square polar channel mapping is in use:
[vorbis_mapping_coupling_steps] = read 8 bits as unsigned integer and add one
for [j] each of [vorbis_mapping_coupling_steps] steps:
vector [vorbis_mapping_magnitude] element [j]= read ilog([audio_
channels] - 1) bits as unsigned integer
vector [vorbis_mapping_angle] element [j]= read ilog([audio_
channels] - 1) bits as unsigned integer
the numbers read in the above two steps are channel numbers representing the channel to treat as magnitude and the channel to treat
as angle, respectively. If for any coupling step the angle channel number equals the magnitude channel number, the magnitude
channel number is greater than [audio_channels]-1, or the angle
28
channel is greater than [audio_channels]-1, the stream is undecodable.

B. if unset, [vorbis_mapping_coupling_steps] = 0
iii. read 2 bits (reserved field); if the value is nonzero, the stream is undecodable
iv. if [vorbis_mapping_submaps] is greater than one, we read channel multiplex settings. For each [j] of [audio_channels] channels:
A. vector [vorbis_mapping_mux] element [j] = read 4 bits as unsigned
integer
B. if the value is greater than the highest numbered submap ([vorbis_
mapping_submaps] - 1), this in an error condition rendering the stream
undecodable
v. for each submap [j] of [vorbis_mapping_submaps] submaps, read the
floor and residue numbers for use in decoding that submap:
A. read and discard 8 bits (the unused time configuration placeholder)
B. read 8 bits as unsigned integer for the floor number; save in vector
[vorbis_mapping_submap_floor] element [j]
C. verify the floor number is not greater than the highest number floor
configured for the bitstream. If it is, the bitstream is undecodable
D. read 8 bits as unsigned integer for the residue number; save in vector
[vorbis_mapping_submap_residue] element [j]
E. verify the residue number is not greater than the highest number residue
configured for the bitstream. If it is, the bitstream is undecodable
vi. save this mapping configuration in slot [i] of the mapping configuration
array [vorbis_mapping_configurations].
Modes
1. [vorbis_mode_count] = read 6 bits as unsigned integer and add one
2. For each of [vorbis_mode_count] mode numbers:
a) [vorbis_mode_blockflag] = read 1 bit
b) [vorbis_mode_windowtype] = read 16 bits as unsigned integer
c) [vorbis_mode_transformtype] = read 16 bits as unsigned integer
d) [vorbis_mode_mapping] = read 8 bits as unsigned integer
29
e) verify ranges; zero is the only legal value in Vorbis I for [vorbis_mode_windowtype]
and [vorbis_mode_transformtype]. [vorbis_mode_mapping] must not be
greater than the highest number mapping in use. Any illegal values render the
stream undecodable.
f) save this mode configuration in slot [i] of the mode configuration array [vorbis_
mode_configurations].
3. read 1 bit as a framing flag. If unset, a framing error occurred and the stream is not
decodable.
After reading mode descriptions, setup header decode is complete.
4.3. Audio packet decode and synthesis

Following the three header packets, all packets in a Vorbis I stream are audio. The first
step of audio packet decode is to read and verify the packet type. A non-audio packet when
audio is expected indicates stream corruption or a non-compliant stream. The decoder must
ignore the packet and not attempt decoding it to audio.
4.3.1. packet type, mode and window decode
1. read 1 bit [packet_type]; check that packet type is 0 (audio)
2. read ilog([vorbis mode count]-1) bits [mode_number]
3. decode blocksize [n] is equal to [blocksize_0] if [vorbis_mode_blockflag] is 0,
else [n] is equal to [blocksize_1].
4. perform window selection and setup; this window is used later by the inverse MDCT:
a) if this is a long window (the [vorbis_mode_blockflag] flag of this mode is
set):
i. read 1 bit for [previous_window_flag]
ii. read 1 bit for [next_window_flag]
iii. if [previous_window_flag] is not set, the left half of the window will be
a hybrid window for lapping with a short block. See Section 1.3.2, Window shape decode (long windows only) for an illustration of overlapping
dissimilar windows. Else, the left half window will have normal long shape.
iv. if [next_window_flag] is not set, the right half of the window will be a
hybrid window for lapping with a short block. See Section 1.3.2, Window shape decode (long windows only) for an illustration of overlapping
dissimilar windows. Else, the left right window will have normal long shape.
30
b) if this is a short window, the window is always the same short-window shape.
Vorbis windows all use the slope function y = sin( 2 sin2 ((x + 0.5)/n )), where n is
window size and x ranges 0 . . . n 1, but dissimilar lapping requirements can affect overall
shape. Window generation proceeds as follows:
1. [window_center] = [n] / 2
2. if ([vorbis_mode_blockflag] is set and [previous_window_flag] is not set) then
a) [left_window_start] = [n]/4 - [blocksize_0]/4
b) [left_window_end] = [n]/4 + [blocksize_0]/4
c) [left_n] = [blocksize_0]/2
else
a) [left_window_start] = 0
b) [left_window_end] = [window_center]
c) [left_n] = [n]/2
3. if ([vorbis_mode_blockflag] is set and [next_window_flag] is not set) then
a) [right_window_start] = [n]*3/4 - [blocksize_0]/4
b) [right_window_end] = [n]*3/4 + [blocksize_0]/4
c) [right_n] = [blocksize_0]/2
else
a) [right_window_start] = [window_center]
b) [right_window_end] = [n]
c) [right_n] = [n]/2
4. window from range 0 ... [left_window_start]-1 inclusive is zero
5. for [i] in range [left_window_start] ... [left_window_end]-1, window([i]) =
sin( 2 sin2 ( ([i]-[left_window_start]+0.5) / [left_n] 2 ) )
6. window from range [left_window_end] ... [right_window_start]-1 inclusive is
one
7. for [i] in range [right_window_start] ... [right_window_end]-1, window([i])
= sin( 2 sin2 ( ([i]-[right_window_start]+0.5) / [right_n] 2 + 2 ) )
8. window from range [right_window_start] ... [n]-1 is zero
31
An end-of-packet condition up to this point should be considered an error that discards

this packet from the stream. An end of packet condition past this point is to be considered
a possible nominal occurrence.
4.3.2. floor curve decode
From this point on, we assume out decode context is using mode number [mode_number]
from configuration array [vorbis_mode_configurations] and the map number [vorbis_
mode_mapping] (specified by the current mode) taken from the mapping configuration
array [vorbis_mapping_configurations].
Floor curves are decoded one-by-one in channel order.
For each floor [i] of [audio_channels]
1. [submap_number] = element [i] of vector [vorbis mapping mux]
2. [floor_number] = element [submap_number] of vector [vorbis submap floor]
3. if the floor type of this floor (vector [vorbis_floor_types] element [floor_number])
is zero then decode the floor for channel [i] according to the subsubsection 6.2.2,
packet decode
4. if the type of this floor is one then decode the floor for channel [i] according to the
subsubsection 7.2.3, packet decode
5. save the needed decoded floor information for channel for later synthesis
6. if the decoded floor returned unused, set vector [no_residue] element [i] to true,
else set vector [no_residue] element [i] to false
An end-of-packet condition during floor decode shall result in packet decode zeroing all
channel output vectors and skipping to the add/overlap output stage.
4.3.3. nonzero vector propagate
A possible result of floor decode is that a specific vector is marked unused which indicates
that that final output vector is all-zero values (and the floor is zero). The residue for that
vector is not coded in the stream, save for one complication. If some vectors are used and
some are not, channel coupling could result in mixing a zeroed and nonzeroed vector to
produce two nonzeroed vectors.
for each [i] from 0 ... [vorbis_mapping_coupling_steps]-1
1. if either [no_residue] entry for channel ([vorbis_mapping_magnitude] element
[i]) or channel ([vorbis_mapping_angle] element [i]) are set to false, then both
32
must be set to false. Note that an unused floor has no decoded floor information;
it is important that this is remembered at floor curve synthesis time.
4.3.4. residue decode
Unlike floors, which are decoded in channel order, the residue vectors are decoded in
submap order.
for each submap [i] in order from 0 ... [vorbis_mapping_submaps]-1
1. [ch] = 0
2. for each channel [j] in order from 0 ... [audio_channels] - 1
a) if channel [j] in submap [i] (vector [vorbis_mapping_mux] element [j] is
equal to [i])
i. if vector [no_residue] element [j] is true
A. vector [do_not_decode_flag] element [ch] is set
else
A. vector [do_not_decode_flag] element [ch] is unset
ii. increment [ch]
3. [residue_number] = vector [vorbis_mapping_submap_residue] element [i]
4. [residue_type] = vector [vorbis_residue_types] element [residue_number]
5. decode [ch] vectors using residue [residue_number], according to type [residue_
type], also passing vector [do_not_decode_flag] to indicate which vectors in the
bundle should not be decoded. Correct per-vector decode length is [n]/2.
6. [ch] = 0
7. for each channel [j] in order from 0 ... [audio_channels]
a) if channel [j] is in submap [i] (vector [vorbis_mapping_mux] element [j] is
equal to [i])
i. residue vector for channel [j] is set to decoded residue vector [ch]
ii. increment [ch]
4.3.5. inverse coupling
for each [i] from [vorbis_mapping_coupling_steps]-1 descending to 0
33
1. [magnitude_vector] = the residue vector for channel (vector [vorbis_mapping_

magnitude] element [i])
2. [angle_vector] = the residue vector for channel (vector [vorbis_mapping_angle]
element [i])
3. for each scalar value [M] in vector [magnitude_vector] and the corresponding scalar
value [A] in vector [angle_vector]:
a) if ([M] is greater than zero)
i. if ([A] is greater than zero)
A. [new_M] = [M]
B. [new_A] = [M]-[A]
else
A. [new_A] = [M]
B. [new_M] = [M]+[A]
else
i. if ([A] is greater than zero)
A. [new_M] = [M]
B. [new_A] = [M]+[A]
else
A. [new_A] = [M]
B. [new_M] = [M]-[A]
b) set scalar value [M] in vector [magnitude_vector] to [new_M]
c) set scalar value [A] in vector [angle_vector] to [new_A]
4.3.6. dot product
For each channel, synthesize the floor curve from the decoded floor information, according
to packet type. Note that the vector synthesis length for floor computation is [n]/2.
For each channel, multiply each element of the floor curve by each element of that channels
residue vector. The result is the dot product of the floor and residue vectors for each
channel; the produced vectors are the length [n]/2 audio spectrum for each channel.
One point is worth mentioning about this dot product; a common mistake in a fixed point
implementation might be to assume that a 32 bit fixed-point representation for floor and
34
residue and direct multiplication of the vectors is sufficient for acceptable spectral depth in
all cases because it happens to mostly work with the current Xiph.Org reference encoder.
However, floor vector values can span 140dB (24 bits unsigned), and the audio spectrum
vector should represent a minimum of 120dB (21 bits with sign), even when output is to
a 16 bit PCM device. For the residue vector to represent full scale if the floor is nailed to
140dB, it must be able to span 0 to +140dB. For the residue vector to reach full scale if
the floor is nailed at 0dB, it must be able to represent 140dB to +0dB. Thus, in order to
handle full range dynamics, a residue vector may span 140dB to +140dB entirely within
spec. A 280dB range is approximately 48 bits with sign; thus the residue vector must be
able to represent a 48 bit range and the dot product must be able to handle an effective 48
bit times 24 bit multiplication. This range may be achieved using large (64 bit or larger)
integers, or implementing a movable binary point representation.
4.3.7. inverse MDCT
Convert the audio spectrum vector of each channel back into time domain PCM audio
via an inverse Modified Discrete Cosine Transform (MDCT). A detailed description of
the MDCT is available in [1]. The window function used for the MDCT is the function
described earlier.
4.3.8. overlap add
Windowed MDCT output is overlapped and added with the right hand data of the previous
window such that the 3/4 point of the previous window is aligned with the 1/4 point of
the current window (as illustrated in Section 1.3.2, Window shape decode (long windows
only)). The overlapped portion produced from overlapping the previous and current frame
data is finished data to be returned by the decoder. This data spans from the center of the
previous window to the center of the current window. In the case of same-sized windows,
the amount of data to return is one-half block consisting of and only of the overlapped
portions. When overlapping a short and long window, much of the returned range does not
actually overlap. This does not damage transform orthogonality. Pay attention however
to returning the correct data range; the amount of data to be returned is:
1
window\_blocksize(previous\_window)/4+window\_blocksize(current\_window)/4
from the center (element windowsize/2) of the previous window to the center (element
windowsize/2-1, inclusive) of the current window.
Data is not returned from the first frame; it must be used to prime the decode engine.
The encoder accounts for this priming when calculating PCM offsets; after the first frame,
the proper PCM output offset is 0 (as no data has been returned yet).
35
4.3.9. output channel order

Vorbis I specifies only a channel mapping type 0. In mapping type 0, channel mapping is implicitly defined as follows for standard audio applications. As of revision 16781 (20100113),
the specification adds defined channel locations for 6.1 and 7.1 surround. Ordering/location
for greater-than-eight channels remains left to the implementation.
These channel orderings refer to order within the encoded stream. It is naturally possible
for a decoder to produce output with channels in any order. Any such decoder should
explicitly document channel reordering behavior.
one channel the stream is monophonic
two channels the stream is stereo. channel order: left, right
three channels the stream is a 1d-surround encoding. channel order: left, center, right
four channels the stream is quadraphonic surround. channel order: front left, front right,
rear left, rear right
five channels the stream is five-channel surround. channel order: front left, center, front
right, rear left, rear right
six channels the stream is 5.1 surround. channel order: front left, center, front right, rear
left, rear right, LFE
seven channels the stream is 6.1 surround. channel order: front left, center, front right,
side left, side right, rear center, LFE
eight channels the stream is 7.1 surround. channel order: front left, center, front right,
side left, side right, rear left, rear right, LFE
greater than eight channels channel use and order is defined by the application
Applications using Vorbis for dedicated purposes may define channel mapping as seen fit.
Future channel mappings (such as three and four channel Ambisonics) will make use of
channel mappings other than mapping 0.
36
5. comment field and header specification

5.1. Overview
The Vorbis text comment header is the second (of three) header packets that begin a Vorbis
bitstream. It is meant for short text comments, not arbitrary metadata; arbitrary metadata
belongs in a separate logical bitstream (usually an XML stream type) that provides greater
structure and machine parseability.
The comment field is meant to be used much like someone jotting a quick note on the
bottom of a CDR. It should be a little information to remember the disc by and explain it
to others; a short, to-the-point text note that need not only be a couple words, but isnt
going to be more than a short paragraph. The essentials, in other words, whatever they
turn out to be, eg:
Honest Bob and the Factory-to-Dealer-Incentives, Im Still Around, opening
for Moxy Fr
uvous, 1997.
5.2. Comment encoding

5.2.1. Structure
The comment header is logically a list of eight-bit-clean vectors; the number of vectors is
bounded to 232 1 and the length of each vector is limited to 232 1 bytes. The vector
length is encoded; the vector contents themselves are not null terminated. In addition to
the vector list, there is a single vector for vendor name (also 8 bit clean, length encoded
in 32 bits). For example, the 1.0 release of libvorbis set the vendor string to Xiph.Org
libVorbis I 20020717.
The vector lengths and number of vectors are stored lsb first, according to the bit packing
conventions of the vorbis codec. However, since data in the comment header is octetaligned, they can simply be read as unaligned 32 bit little endian unsigned integers.
The comment header is decoded as follows:
1
2
3
4
5
6
7
8
9
10
1)
2)
3)
4)
[vendor\_length] = read an unsigned integer of 32 bits

[vendor\_string] = read a UTF-8 vector as [vendor\_length] octets
[user\_comment\_list\_length] = read an unsigned integer of 32 bits
iterate [user\_comment\_list\_length] times {
5) [length] = read an unsigned integer of 32 bits
6) this iterations user comment = read a UTF-8 vector as [length] octets
}
7) [framing\_bit] = read a single bit as boolean
8) if ( [framing\_bit] unset or end-of-packet ) then ERROR
9) done.
37
5.2.2. Content vector format

The comment vectors are structured similarly to a UNIX environment variable. That is,
comment fields consist of a field name and a corresponding value and look like:
1
2
comment[0]="ARTIST=me";
comment[1]="TITLE=the sound of Vorbis";
The field name is case-insensitive and may consist of ASCII 0x20 through 0x7D, 0x3D
(=) excluded. ASCII 0x41 through 0x5A inclusive (characters A-Z) is to be considered
equivalent to ASCII 0x61 through 0x7A inclusive (characters a-z).
The field name is immediately followed by ASCII 0x3D (=); this equals sign is used to
terminate the field name.
0x3D is followed by 8 bit clean UTF-8 encoded value of the field contents to the end of the
field.
Field names Below is a proposed, minimal list of standard field names with a description
of intended use. No single or group of field names is mandatory; a comment header may
contain one, all or none of the names in this list.
TITLE Track/Work name
VERSION The version field may be used to differentiate multiple versions of the same
track title in a single collection. (e.g. remix info)
ALBUM The collection name to which this track belongs
TRACKNUMBER The track number of this piece if part of a specific larger collection or
album
ARTIST The artist generally considered responsible for the work. In popular music this is
usually the performing band or singer. For classical music it would be the composer.
For an audio book it would be the author of the original text.
PERFORMER The artist(s) who performed the work. In classical music this would be the
conductor, orchestra, soloists. In an audio book it would be the actor who did the
reading. In popular music this is typically the same as the ARTIST and is omitted.
COPYRIGHT Copyright attribution, e.g., 2001 Nobodys Band or 1999 Jack Moffitt
LICENSE License information, eg, All Rights Reserved, Any Use Permitted, a URL to a
license such as a Creative Commons license (www.creativecommons.org/blahblah/license.html)
or the EFF Open Audio License (distributed under the terms of the Open Audio
License. see http://www.eff.org/IP/Open licenses/eff oal.html for details), etc.
ORGANIZATION Name of the organization producing the track (i.e. the record label)
DESCRIPTION A short text description of the contents
38
GENRE A short text indication of music genre

DATE Date the track was recorded
LOCATION Location where track was recorded
CONTACT Contact information for the creators or distributors of the track. This could
be a URL, an email address, the physical address of the producing label.
ISRC International Standard Recording Code for the track; see the ISRC intro page for
more information on ISRC numbers.
Implications Field names should not be internationalized; this is a concession to simplicity not an attempt to exclude the majority of the world that doesnt speak English.
Field contents, however, use the UTF-8 character encoding to allow easy representation of
any language.
We have the length of the entirety of the field and restrictions on the field name so that the
field name is bounded in a known way. Thus we also have the length of the field contents.
Individual vendors may use non-standard field names within reason. The proper use of
comment fields should be clear through context at this point. Abuse will be discouraged.
There is no vendor-specific prefix to nonstandard field names. Vendors should make some
effort to avoid arbitrarily polluting the common namespace. We will generally collect the
more useful tags here to help with standardization.
Field names are not required to be unique (occur once) within a comment header. As
an example, assume a track was recorded by three well know artists; the following is
permissible, and encouraged:
1
2
3
ARTIST=Dizzy Gillespie
ARTIST=Sonny Rollins
ARTIST=Sonny Stitt
5.2.3. Encoding
The comment header comprises the entirety of the second bitstream header packet. Unlike
the first bitstream header packet, it is not generally the only packet on the second page
and may not be restricted to within the second bitstream page. The length of the comment
header packet is (practically) unbounded. The comment header packet is not optional; it
must be present in the bitstream even if it is effectively empty.
The comment header is encoded as follows (as per Oggs standard bitstream mapping which
renders least-significant-bit of the word to be coded into the least significant available bit
of the current bitstream octet first):
1. Vendor string length (32 bit unsigned quantity specifying number of octets)
39
2. Vendor string ([vendor string length] octets coded from beginning of string to end of
string, not null terminated)
3. Number of comment fields (32 bit unsigned quantity specifying number of fields)
4. Comment field 0 length (if [Number of comment fields] > 0; 32 bit unsigned quantity
specifying number of octets)
5. Comment field 0 ([Comment field 0 length] octets coded from beginning of string to
end of string, not null terminated)
6. Comment field 1 length (if [Number of comment fields] > 1...)...
This is actually somewhat easier to describe in code; implementation of the above can be
found in vorbis/lib/info.c, _vorbis_pack_comment() and _vorbis_unpack_comment().
40

6.1. Overview
Vorbis floor type zero uses Line Spectral Pair (LSP, also alternately known as Line Spectral Frequency or LSF) representation to encode a smooth spectral envelope curve as the
frequency response of the LSP filter. This representation is equivalent to a traditional
all-pole infinite impulse response filter as would be used in linear predictive coding; LSP
representation may be converted to LPC representation and vice-versa.
6.2. Floor 0 format

Floor zero configuration consists of six integer fields and a list of VQ codebooks for use in
coding/decoding the LSP filter coefficient values used by each frame.
6.2.1. header decode
Configuration information for instances of floor zero decodes from the codec setup header
(third packet). configuration decode proceeds as follows:
1
2
3
4
5
6
7
1)
2)
3)
4)
5)
6)
7)
[floor0_order] = read an unsigned integer of 8 bits

[floor0_rate] = read an unsigned integer of 16 bits
[floor0_bark_map_size] = read an unsigned integer of 16 bits
[floor0_amplitude_bits] = read an unsigned integer of six bits
[floor0_amplitude_offset] = read an unsigned integer of eight bits
[floor0_number_of_books] = read an unsigned integer of four bits and add 1
array [floor0_book_list] = read a list of [floor0_number_of_books] unsigned integers of eight bits each;
An end-of-packet condition during any of these bitstream reads renders this stream undecodable. In addition, any element of the array [floor0_book_list] that is greater than
the maximum codebook number for this bitstream is an error condition that also renders
the stream undecodable.
6.2.2. packet decode
Extracting a floor0 curve from an audio packet consists of first decoding the curve amplitude
and [floor0_order] LSP coefficient values from the bitstream, and then computing the
floor curve, which is defined as the frequency response of the decoded LSP filter.
Packet decode proceeds as follows:
1
2
3
4
5
1) [amplitude] = read an unsigned integer of [floor0_amplitude_bits] bits

2) if ( [amplitude] is greater than zero ) {
3) [coefficients] is an empty, zero length vector
4) [booknumber] = read an unsigned integer of ilog( [floor0_number_of_books] ) bits
5) if ( [booknumber] is greater than the highest number decode codebook ) then packet is undecodable
41
6
7
8
9
10
11
6)
7)
8)
9)
10)
11)
[last] = zero;
vector [temp_vector] = read vector from bitstream using codebook number [floor0_book_list] element [booknumber] in
add the scalar value [last] to each scalar in vector [temp_vector]
[last] = the value of the last scalar in vector [temp_vector]
concatenate [temp_vector] onto the end of the [coefficients] vector
if (length of vector [coefficients] is less than [floor0_order], continue at step 6
12
13
14
15
12) done.
16
Take note of the following properties of decode:

An [amplitude] value of zero must result in a return code that indicates this channel
is unused in this frame (the output of the channel will be all-zeroes in synthesis).
Several later stages of decode dont occur for an unused channel.
An end-of-packet condition during decode should be considered a nominal occruence;
if end-of-packet is reached during any read operation above, floor decode is to return
unused status as if the [amplitude] value had read zero at the beginning of decode.
The book number used for decode can, in fact, be stored in the bitstream in ilog(
[floor0_number_of_books] - 1 ) bits. Nevertheless, the above specification is correct and values greater than the maximum possible book value are reserved.
The number of scalars read into the vector [coefficients] may be greater than
[floor0_order], the number actually required for curve computation. For example,
if the VQ codebook used for the floor currently being decoded has a [codebook_
dimensions] value of three and [floor0_order] is ten, the only way to fill all the
needed scalars in [coefficients] is to to read a total of twelve scalars as four vectors
of three scalars each. This is not an error condition, and care must be taken not to
allow a buffer overflow in decode. The extra values are not used and may be ignored
or discarded.
6.2.3. curve computation
Given an [amplitude] integer and [coefficients] vector from packet decode as well as
the [floor0 order], [floor0 rate], [floor0 bark map size], [floor0 amplitude bits] and [floor0
amplitude offset] values from floor setup, and an output vector size [n] specified by the
decode process, we compute a floor output vector.
If the value [amplitude] is zero, the return value is a length [n] vector with all-zero
scalars. Otherwise, begin by assuming the following definitions for the given vector to be
synthesized:
(
mapi =
min(floor0_bark_map_size 1, f oobar) for i [0, n 1]

1
for i = n
42
where
$
floor0_bark_map_size
floor0_rate i
f oobar = bark
2n
bark(.5 floor0_rate)

and
bark(x) = 13.1 arctan(.00074x) + 2.24 arctan(.0000000185x2 + .0001x)
The above is used to synthesize the LSP curve on a Bark-scale frequency axis, then map
the result to a linear-scale frequency axis. Similarly, the below calculation synthesizes the
output LSP curve [output] on a log (dB) amplitude scale, mapping it to linear amplitude
in the last step:
1. [i] = 0
2. [] = * map element [i] / [floor0_bark_map_size]
3. if ( [floor0_order] is odd )
a) calculate [p] and [q] according to:
_order3
floor0
2
Y
p = (1 cos )
4(cos([coefficients]2j+1 ) cos )2
j=0
_order1
floor0
q =
1
4
2
Y
4(cos([coefficients]2j ) cos )2
j=0
else [floor0_order] is even

b) calculate [p] and [q] according to:
_order2
floor0
(1 cos )
p =
2
2
Y
4(cos([coefficients]2j+1 ) cos )2
j=0
_order2
floor0
(1 + cos )
q =
2
2
Y
4(cos([coefficients]2j ) cos )2
j=0
4. calculate [linear_floor_value] according to:

!!
amplitude floor0_amplitute_offset
exp .11512925
floor0_amplitude_offset
(2floor0_amplitude_bits 1) p + q
5. [iteration_condition] = map element [i]
43
6. [output] element [i] = [linear_floor_value]

7. increment [i]
8. if ( map element [i] is equal to [iteration_condition] ) continue at step 5
9. if ( [i] is less than [n] ) continue at step 2
10. done
44

7.1. Overview
Vorbis floor type one uses a piecewise straight-line representation to encode a spectral
envelope curve. The representation plots this curve mechanically on a linear frequency
axis and a logarithmic (dB) amplitude axis. The integer plotting algorithm used is similar
to Bresenhams algorithm.
7.2. Floor 1 format

7.2.1. model
Floor type one represents a spectral curve as a series of line segments. Synthesis constructs
a floor curve using iterative prediction in a process roughly equivalent to the following
simplified description:
the first line segment (base case) is a logical line spanning from x 0,y 0 to x 1,y 1
where in the base case x 0=0 and x 1=[n], the full range of the spectral floor to be
computed.
the induction step chooses a point x new within an existing logical line segment and
produces a y new value at that point computed from the existing lines y value at
x new (as plotted by the line) and a difference value decoded from the bitstream
packet.
floor computation produces two new line segments, one running from x 0,y 0 to x
new,y new and from x new,y new to x 1,y 1. This step is performed logically even if
y new represents no change to the amplitude value at x new so that later refinement
is additionally bounded at x new.
the induction step repeats, using a list of x values specified in the codec setup header
at floor 1 initialization time. Computation is completed at the end of the x value
list.
Consider the following example, with values chosen for ease of understanding rather than
representing typical configuration:
For the below example, we assume a floor setup with an [n] of 128. The list of selected
X values in increasing order is 0,16,32,48,64,80,96,112 and 128. In list order, the values
interleave as 0, 128, 64, 32, 96, 16, 48, 80 and 112. The corresponding list-order Y values as
decoded from an example packet are 110, 20, -5, -45, 0, -25, -10, 30 and -10. We compute
the floor in the following way, beginning with the first line:
45
Figure 7: graph of example floor

We now draw new logical lines to reflect the correction to new Y, and iterate for X positions
32 and 96:

Although the new Y value at X position 96 is unchanged, it is still used later as an endpoint
for further refinement. From here on, the pattern should be clear; we complete the floor
computation as follows:
46

A more efficient algorithm with carefully defined integer rounding behavior is used for
actual decode, as described later. The actual algorithm splits Y value computation and
line plotting into two steps with modifications to the above algorithm to eliminate noise
accumulation through integer roundoff/truncation.
A list of floor X values is stored in the packet header in interleaved format (used in list order
during packet decode and synthesis). This list is split into partitions, and each partition
is assigned to a partition class. X positions 0 and [n] are implicit and do not belong to an
explicit partition or partition class.
A partition class consists of a representation vector width (the number of Y values which
the partition class encodes at once), a subclass value representing the number of alternate
entropy books the partition class may use in representing Y values, the list of [subclass]
books and a master book used to encode which alternate books were chosen for representation in a given packet. The master/subclass mechanism is meant to be used as a flexible
representation cascade while still using codebooks only in a scalar context.
1
2
3
4
1) [floor1_partitions] = read 5 bits as unsigned integer

2) [maximum_class] = -1
3) iterate [i] over the range 0 ... [floor1_partitions]-1 {
4) vector [floor1_partition_class_list] element [i] = read 4 bits as unsigned integer
6
7
8
9
10
11
5) [maximum_class] = largest integer scalar value in vector [floor1_partition_class_list]

6) iterate [i] over the range 0 ... [maximum_class] {
12
13
14
15
7) vector [floor1_class_dimensions] element [i] = read 3 bits as unsigned integer and add 1
8) vector [floor1_class_subclasses] element [i] = read 2 bits as unsigned integer
9) if ( vector [floor1_class_subclasses] element [i] is nonzero ) {
16
10) vector [floor1_class_masterbooks] element [i] = read 8 bits as unsigned integer
17
18
19
47
20
11) iterate [j] over the range 0 ... (2 exponent [floor1_class_subclasses] element [i]) - 1 {
21
22
12) array [floor1_subclass_books] element [i],[j] =

read 8 bits as unsigned integer and subtract one
23
24
25
26
27
28
29
30
31
32
33
13)
14)
15)
16)
17)
18)
[floor1_multiplier] = read 2 bits as unsigned integer and add one

[rangebits] = read 4 bits as unsigned integer
vector [floor1_X_list] element [0] = 0
vector [floor1_X_list] element [1] = 2 exponent [rangebits];
[floor1_values] = 2
iterate [i] over the range 0 ... [floor1_partitions]-1 {
34
19) [current_class_number] = vector [floor1_partition_class_list] element [i]

20) iterate [j] over the range 0 ... ([floor1_class_dimensions] element [current_class_number])-1 {
21) vector [floor1_X_list] element ([floor1_values]) =
read [rangebits] bits as unsigned integer
22) increment [floor1_values] by one
}
35
36
37
38
39
40
41
42
43
23) done
An end-of-packet condition while reading any aspect of a floor 1 configuration during setup
renders a stream undecodable. In addition, a [floor1_class_masterbooks] or [floor1_
subclass_books] scalar element greater than the highest numbered codebook configured
in this stream is an error condition that renders the stream undecodable. Vector [floor1
x list] is limited to a maximum length of 65 elements; a setup indicating more than 65
total elements (including elements 0 and 1 set prior to the read loop) renders the stream
undecodable. All vector [floor1 x list] element values must be unique within the vector; a
non-unique value renders the stream undecodable.
Packet decode begins by checking the [nonzero] flag:
1
1) [nonzero] = read 1 bit as boolean
If [nonzero] is unset, that indicates this channel contained no audio energy in this frame.
Decode immediately returns a status indicating this floor curve (and thus this channel) is
unused this frame. (A return status of unused is different from decoding a floor that has
all points set to minimum representation amplitude, which happens to be approximately
-140dB).
Assuming [nonzero] is set, decode proceeds as follows:
1
2
3
4
5
1)
2)
3)
4)
5)
[range] = vector { 256, 128, 86, 64 } element ([floor1_multiplier]-1)

vector [floor1_Y] element [0] = read ilog([range]-1) bits as unsigned integer
vector [floor1_Y] element [1] = read ilog([range]-1) bits as unsigned integer
[offset] = 2;
iterate [i] over the range 0 ... [floor1_partitions]-1 {
6
7
6) [class] = vector [floor1_partition_class]
element [i]
48
8
9
10
11
12
7)
8)
9)
10)
11)
[cdim] = vector [floor1_class_dimensions] element [class]

[cbits] = vector [floor1_class_subclasses] element [class]
[csub] = (2 exponent [cbits])-1
[cval] = 0
if ( [cbits] is greater than zero ) {
13
12) [cval] = read from packet using codebook number

(vector [floor1_class_masterbooks] element [class]) in scalar context
14
15
16
17
18
13) iterate [j] over the range 0 ... [cdim]-1 {
19
14) [book] = array [floor1_subclass_books] element [class],([cval] bitwise AND [csub])

15) [cval] = [cval] right shifted [cbits] bits
16) if ( [book] is not less than zero ) {
20
21
22
23
17) vector [floor1_Y] element ([j]+[offset]) = read from packet using codebook
[book] in scalar context
24
25
26
} else [book] is less than zero {
27
28
18) vector [floor1_Y] element ([j]+[offset]) = 0
29
30
31
32
33
34
19) [offset] = [offset] + [cdim]
35
36
37
38
20) done
An end-of-packet condition during curve decode should be considered a nominal occurrence;

if end-of-packet is reached during any read operation above, floor decode is to return
unused status as if the [nonzero] flag had been unset at the beginning of decode.
Vector [floor1_Y] contains the values from packet decode needed for floor 1 synthesis.
7.2.4. curve computation
Curve computation is split into two logical steps; the first step derives final Y amplitude
values from the encoded, wrapped difference values taken from the bitstream. The second
step plots the curve lines. Also, although zero-difference values are used in the iterative
prediction to find final Y values, these points are conditionally skipped during final line
computation in step two. Skipping zero-difference values allows a smoother line fit.
Although some aspects of the below algorithm look like inconsequential optimizations,
implementors are warned to follow the details closely. Deviation from implementing a
strictly equivalent algorithm can result in serious decoding errors.
Additional note: Although [floor1_final_Y] values in the prediction loop and at the end
of step 1 are inherently limited by the prediction algorithm to [0, [range]), it is possible
to abuse the setup and codebook machinery to produce negative or over-range results.
We suggest that decoder implementations guard the values in vector [floor1_final_Y]
49
by clamping each element to [0, [range]) after step 1. Variants of this suggestion are
acceptable as valid floor1 setups cannot produce out of range values.
step 1: amplitude value synthesis Unwrap the always-positive-or-zero values read from
the packet into +/- difference values, then apply to line prediction.
1
2
3
4
5
6
1)
2)
3)
4)
5)
6)
[range] = vector { 256, 128, 86, 64 } element ([floor1_multiplier]-1)

vector [floor1_step2_flag] element [0] = set
vector [floor1_step2_flag] element [1] = set
vector [floor1_final_Y] element [0] = vector [floor1_Y] element [0]
vector [floor1_final_Y] element [1] = vector [floor1_Y] element [1]
iterate [i] over the range 2 ... [floor1_values]-1 {
7
8
9
7) [low_neighbor_offset] = low_neighbor([floor1_X_list],[i])
8) [high_neighbor_offset] = high_neighbor([floor1_X_list],[i])
10
11
12
13
14
15
9) [predicted] = render_point( vector [floor1_X_list] element [low_neighbor_offset],

vector [floor1_final_Y] element [low_neighbor_offset],
vector [floor1_X_list] element [high_neighbor_offset],
vector [floor1_final_Y] element [high_neighbor_offset],
vector [floor1_X_list] element [i] )
16
17
18
19
20
10)
11)
12)
13)
[val] = vector [floor1_Y] element [i]

[highroom] = [range] - [predicted]
[lowroom] = [predicted]
if ( [highroom] is less than [lowroom] ) {
21
14) [room] = [highroom] * 2
22
23
24
} else [highroom] is not less than [lowroom] {
25
15) [room] = [lowroom] * 2
26
27
28
29
30
16) if ( [val] is nonzero ) {
31
32
33
34
35
17)
18)
19)
20)
vector [floor1_step2_flag]
if ( [val] is greater than
element [low_neighbor_offset] = set

element [high_neighbor_offset] = set
element [i] = set
or equal to [room] ) {
36
37
21) if ( [highroom] is greater than [lowroom] ) {
38
22) vector [floor1_final_Y] element [i] = [val] - [lowroom] + [predicted]
39
40
41
} else [highroom] is not greater than [lowroom] {
42
23) vector [floor1_final_Y] element [i] = [predicted] - [val] + [highroom] - 1
43
44
45
46
47
} else [val] is less than [room] {
48
49
24) if ([val] is odd) {
50
51
52
25) vector [floor1_final_Y] element [i] =

[predicted] - (([val] + 1) divided by
2 using integer division)
53
54
} else [val] is even {
55
56
57
26) vector [floor1_final_Y] element [i] =

[predicted] + ([val] / 2 using integer division)
58
50
59
60
61
62
} else [val] is zero {
63
64
27) vector [floor1_step2_flag] element [i] = unset

28) vector [floor1_final_Y] element [i] = [predicted]
65
66
67
68
69
70
71
72
29) done
73
step 2: curve synthesis Curve synthesis generates a return vector [floor] of length [n]
(where [n] is provided by the decode process calling to floor decode). Floor 1 curve
synthesis makes use of the [floor1_X_list], [floor1_final_Y] and [floor1_
step2_flag] vectors, as well as [floor1 multiplier] and [floor1 values] values.
Decode begins by sorting the scalars from vectors [floor1_X_list], [floor1_final_
Y] and [floor1_step2_flag] together into new vectors [floor1_X_list], [floor1_
final_Y] and [floor1_step2_flag] according to ascending sort order of the values in [floor1_X_list]. That is, sort the values of [floor1_X_list] and then
apply the same permutation to elements of the other two vectors so that the X, Y
and step2 flag values still match.
Then compute the final curve in one pass:
1
2
3
4
1)
2)
3)
4)
[hx] = 0
[lx] = 0
[ly] = vector [floor1_final_Y] element [0] * [floor1_multiplier]
iterate [i] over the range 1 ... [floor1_values]-1 {
5) if ( [floor1_step2_flag] element [i] is set ) {
6
7
6) [hy] = [floor1_final_Y] element [i] * [floor1_multiplier]

7) [hx] = [floor1_X_list] element [i]
8) render_line( [lx], [ly], [hx], [hy], [floor] )
9) [lx] = [hx]
10) [ly] = [hy]
}
8
9
10
11
12
13
14
15
16
11) if ( [hx] is less than [n] ) {
17
12) render_line( [hx], [hy], [n], [hy], [floor] )
18
19
20
21
22
13) if ( [hx] is greater than [n] ) {
23
14) truncate vector [floor] to [n] elements
24
25
26
27
28
29
15) for each scalar in vector [floor], perform a lookup substitution using
the scalar value from [floor] as an offset into the vector [floor1_inverse_dB_static_table]
30
51
31
16) done
32
52
8. Residue setup and decode

8.1. Overview
A residue vector represents the fine detail of the audio spectrum of one channel in an audio
frame after the encoder subtracts the floor curve and performs any channel coupling. A
residue vector may represent spectral lines, spectral magnitude, spectral phase or hybrids
as mixed by channel coupling. The exact semantic content of the vector does not matter
to the residue abstraction.
Whatever the exact qualities, the Vorbis residue abstraction codes the residue vectors into
the bitstream packet, and then reconstructs the vectors during decode. Vorbis makes use of
three different encoding variants (numbered 0, 1 and 2) of the same basic vector encoding
abstraction.
8.2. Residue format

Residue format partitions each vector in the vector bundle into chunks, classifies each
chunk, encodes the chunk classifications and finally encodes the chunks themselves using the
the specific VQ arrangement defined for each selected classification. The exact interleaving
and partitioning vary by residue encoding number, however the high-level process used to
classify and encode the residue vector is the same in all three variants.
A set of coded residue vectors are all of the same length. High level coding structure,
ignoring for the moment exactly how a partition is encoded and simply trusting that it is,
is as follows:
Each vector is partitioned into multiple equal sized chunks according to configuration
specified. If we have a vector size of n, a partition size residue partition size, and a
total of ch residue vectors, the total number of partitioned chunks coded is n/residue
partition size*ch. It is important to note that the integer division truncates. In the
below example, we assume an example residue partition size of 8.
Each partition in each vector has a classification number that specifies which of
multiple configured VQ codebook setups are used to decode that partition. The
classification numbers of each partition can be thought of as forming a vector in
their own right, as in the illustration below. Just as the residue vectors are coded
in grouped partitions to increase encoding efficiency, the classification vector is also
partitioned into chunks. The integer elements of each scalar in a classification chunk
are built into a single scalar that represents the classification numbers in that chunk.
In the below example, the classification codeword encodes two classification numbers.
The values in a residue vector may be encoded monolithically in a single pass through
the residue vector, but more often efficient codebook design dictates that each vector
53
is encoded as the additive sum of several passes through the residue vector using more
than one VQ codebook. Thus, each residue value potentially accumulates values
from multiple decode passes. The classification value associated with a partition is
the same in each pass, thus the classification codeword is coded only in the first pass.
54
Figure 11: illustration of residue vector format
55
8.3. residue 0
Residue 0 and 1 differ only in the way the values within a residue partition are interleaved
during partition encoding (visually treated as a black boxor cyan box or brown boxin
the above figure).
Residue encoding 0 interleaves VQ encoding according to the dimension of the codebook
used to encode a partition in a specific pass. The dimension of the codebook need not be
the same in multiple passes, however the partition size must be an even multiple of the
codebook dimension.
As an example, assume a partition vector of size eight, to be encoded by residue 0 using
codebook sizes of 8, 4, 2 and 1:
1
2
original residue vector: [ 0 1 2 3 4 5 6 7 ]
3
4
codebook dimensions = 8
encoded as: [ 0 1 2 3 4 5 6 7 ]
encoded as: [ 0 2 4 6 ], [ 1 3 5 7 ]
encoded as: [ 0 4 ], [ 1 5 ], [ 2 6 ], [ 3 7 ]
encoded as: [ 0 ], [ 1 ], [ 2 ], [ 3 ], [ 4 ], [ 5 ], [ 6 ], [ 7 ]
5
6
7
8
9
10
11
It is worth mentioning at this point that no configurable value in the residue coding setup
is restricted to a power of two.
8.4. residue 1
Residue 1 does not interleave VQ encoding. It represents partition vector scalars in order.
As with residue 0, however, partition length must be an integer multiple of the codebook
dimension, although dimension may vary from pass to pass.
As an example, assume a partition vector of size eight, to be encoded by residue 0 using
codebook sizes of 8, 4, 2 and 1:
1
2
original residue vector: [ 0 1 2 3 4 5 6 7 ]
3
4
encoded as: [ 0 1 2 3 4 5 6 7 ]
encoded as: [ 0 1 2 3 ], [ 4 5 6 7 ]
encoded as: [ 0 1 ], [ 2 3 ], [ 4 5 ], [ 6 7 ]
encoded as: [ 0 ], [ 1 ], [ 2 ], [ 3 ], [ 4 ], [ 5 ], [ 6 ], [ 7 ]
5
6
7
8
9
10
11
56
8.5. residue 2
Residue type two can be thought of as a variant of residue type 1. Rather than encoding
multiple passed-in vectors as in residue type 1, the ch passed in vectors of length n are
first interleaved and flattened into a single vector of length ch*n. Encoding then proceeds
as in type 1. Decoding is as in type 1 with decode interleave reversed. If operating on a
single vector to begin with, residue type 1 and type 2 are equivalent.
Figure 12: illustration of residue type 2
57
8.6. Residue decode

Header decode for all three residue types is identical.
1
2
3
4
5
1)
2)
3)
4)
5)
[residue\_begin] = read 24 bits as unsigned integer

[residue\_end] = read 24 bits as unsigned integer
[residue\_partition\_size] = read 24 bits as unsigned integer and add one
[residue\_classifications] = read 6 bits as unsigned integer and add one
[residue\_classbook] = read 8 bits as unsigned integer
[residue_begin] and [residue_end] select the specific sub-portion of each vector that
is actually coded; it implements akin to a bandpass where, for coding purposes, the vector effectively begins at element [residue_begin] and ends at [residue_end]. Preceding and following values in the unpacked vectors are zeroed. Note that for residue
type 2, these values as well as [residue_partition_size]apply to the interleaved vector,
not the individual vectors before interleave. [residue_partition_size] is as explained
above, [residue_classifications] is the number of possible classification to which a
partition can belong and [residue_classbook] is the codebook number used to code
classification codewords. The number of dimensions in book [residue_classbook] determines how many classification values are grouped into a single classification codeword.
Note that the number of entries and dimensions in book [residue_classbook], along
with [residue_classifications], overdetermines to possible number of classification
codewords. If [residue_classifications][residue_classbook].dimensions exceeds
[residue_classbook].entries, the bitstream should be regarded to be undecodable.
Next we read a bitmap pattern that specifies which partition classes code values in which
passes.
1
1) iterate [i] over the range 0 ... [residue\_classifications]-1 {
2
3
4
5
6
7
8
9
2)
3)
4)
5)
6)
[high\_bits] = 0
[low\_bits] = read 3 bits as unsigned integer
[bitflag] = read one bit as boolean
if ( [bitflag] is set ) then [high\_bits] = read five bits as unsigned integer
vector [residue\_cascade] element [i] = [high\_bits] * 8 + [low\_bits]
}
7) done
Finally, we read in a list of book numbers, each corresponding to specific bit set in the
cascade bitmap. We loop over the possible codebook classifications and the maximum
possible number of encoding stages (8 in Vorbis I, as constrained by the elements of the
cascade bitmap being eight bits):
1
1) iterate [i] over the range 0 ... [residue\_classifications]-1 {
2
3
2) iterate [j] over the range 0 ... 7 {
4
5
3) if ( vector [residue\_cascade] element [i] bit [j] is set ) {
6
7
4) array [residue\_books] element [i][j] = read 8 bits as unsigned integer
58
} else {
9
10
5) array [residue\_books] element [i][j] = unused
11
12
13
14
15
16
17
6) done
An end-of-packet condition at any point in header decode renders the stream undecodable.
In addition, any codebook number greater than the maximum numbered codebook set up
in this stream also renders the stream undecodable. All codebooks in array [residue books]
are required to have a value mapping. The presence of codebook in array [residue books]
without a value mapping (maptype equals zero) renders the stream undecodable.
Format 0 and 1 packet decode is identical except for specific partition interleave. Format
2 packet decode can be built out of the format 1 decode process. Thus we describe first
the decode infrastructure identical to all three formats.
In addition to configuration information, the residue decode process is passed the number
of vectors in the submap bundle and a vector of flags indicating if any of the vectors are not
to be decoded. If the passed in number of vectors is 3 and vector number 1 is marked do
not decode, decode skips vector 1 during the decode loop. However, even do not decode
vectors are allocated and zeroed.
Depending on the values of [residue_begin] and [residue_end], it is obvious that the
encoded portion of a residue vector may be the entire possible residue vector or some
other strict subset of the actual residue vector size with zero padding at either uncoded
end. However, it is also possible to set [residue_begin] and [residue_end] to specify
a range partially or wholly beyond the maximum vector size. Before beginning residue
decode, limit [residue_begin] and [residue_end] to the maximum possible vector size
as follows. We assume that the number of vectors being encoded, [ch] is provided by the
higher level decoding process.
1
2
3
4
5
1) [actual\_size] = current blocksize/2;

2) if residue encoding is format 2
3) [actual\_size] = [actual\_size] * [ch];
4) [limit\_residue\_begin] = maximum of ([residue\_begin],[actual\_size]);
5) [limit\_residue\_end] = maximum of ([residue\_end],[actual\_size]);
The following convenience values are conceptually useful to clarifying the decode process:
1
2
3
1) [classwords\_per\_codeword] = [codebook\_dimensions] value of codebook [residue\_classbook]

2) [n\_to\_read] = [limit\_residue\_end] - [limit\_residue\_begin]
3) [partitions\_to\_read] = [n\_to\_read] / [residue\_partition\_size]
Packet decode proceeds as follows, matching the description offered earlier in the document.
59
1
2
3
1) allocate and zero all vectors that will be returned.

2) if ([n\_to\_read] is zero), stop; there is no residue to decode.
3) iterate [pass] over the range 0 ... 7 {
4) [partition\_count] = 0
5
6
5) while [partition\_count] is less than [partitions\_to\_read]
7
8
6) if ([pass] is zero) {
9
10
7) iterate [j] over the range 0 .. [ch]-1 {
11
12
8) if vector [j] is not marked do not decode {
13
14
9) [temp] = read from packet using codebook [residue\_classbook] in scalar context

10) iterate [i] descending over the range [classwords\_per\_codeword]-1 ... 0 {
15
16
17
11) array [classifications] element [j],([i]+[partition\_count]) =

[temp] integer modulo [residue\_classifications]
12) [temp] = [temp] / [residue\_classifications] using integer division
18
19
20
21
22
23
24
25
26
27
28
29
13) iterate [i] over the range 0 .. ([classwords\_per\_codeword] - 1) while [partition\_count]

is also less than [partitions\_to\_read] {
30
31
32
14) iterate [j] over the range 0 .. [ch]-1 {
33
34
15) if vector [j] is not marked do not decode {
35
36
16) [vqclass] = array [classifications] element [j],[partition\_count]

17) [vqbook] = array [residue\_books] element [vqclass],[pass]
18) if ([vqbook] is not unused) {
37
38
39
40
19) decode partition into output vector number [j], starting at scalar
offset [limit\_residue\_begin]+[partition\_count]*[residue\_partition\_size] using
codebook number [vqbook] in VQ context
41
42
43
44
45
46
20) increment [partition\_count] by one
47
48
49
50
51
52
53
21) done
54
An end-of-packet condition during packet decode is to be considered a nominal occurrence.

Decode returns the result of vector decode up to that point.
60

Format zero decodes partitions exactly as described earlier in the Residue Format: residue
0 section. The following pseudocode presents the same algorithm. Assume:
[n] is the value in [residue_partition_size]
[v] is the residue vector
[offset] is the beginning read offset in [v]
1
2
1) [step] = [n] / [codebook\_dimensions]

2) iterate [i] over the range 0 ... [step]-1 {
3) vector [entry\_temp] = read vector from packet using current codebook in VQ context
4) iterate [j] over the range 0 ... [codebook\_dimensions]-1 {
4
5
6
5) vector [v] element ([offset]+[i]+[j]*[step]) =

vector [v] element ([offset]+[i]+[j]*[step]) +
vector [entry\_temp] element [j]
7
8
9
10
11
12
13
14
15
6) done
16

Format 1 decodes partitions exactly as described earlier in the Residue Format: residue
1 section. The following pseudocode presents the same algorithm. Assume:
[n] is the value in [residue_partition_size]
[v] is the residue vector
[offset] is the beginning read offset in [v]
1
2
3
1) [i] = 0
2) vector [entry\_temp] = read vector from packet using current codebook in VQ context
3) iterate [j] over the range 0 ... [codebook\_dimensions]-1 {
4) vector [v] element ([offset]+[i]) =

vector [v] element ([offset]+[i]) +
vector [entry\_temp] element [j]
5) increment [i]
5
6
7
8
9
10
11
12
13
6) if ( [i] is less than [n] ) continue at step 2

7) done
61

Format 2 is reducible to format 1. It may be implemented as an additional step prior to
and an additional post-decode step after a normal format 1 decode.
Format 2 handles do not decode vectors differently than residue 0 or 1; if all vectors are
marked do not decode, no decode occurrs. However, if at least one vector is to be decoded,
all the vectors are decoded. We then request normal format 1 to decode a single vector
representing all output channels, rather than a vector for each channel. After decode,
deinterleave the vector into independent vectors, one for each output channel. That is:
1. If all vectors 0 through ch-1 are marked do not decode, allocate and clear a single
vector [v]of length ch*n and skip step 2 below; proceed directly to the post-decode
step.
2. Rather than performing format 1 decode to produce ch vectors of length n each, call
format 1 decode to produce a single vector [v] of length ch*n.
3. Post decode: Deinterleave the single vector [v] returned by format 1 decode as
described above into ch independent vectors, one for each outputchannel, according
to:
1
1) iterate [i] over the range 0 ... [n]-1 {
2) iterate [j] over the range 0 ... [ch]-1 {
3
4
3) output vector number [j] element [i] = vector [v] element ([i] * [ch] + [j])
5
6
7
8
9
10
4) done
62
9. Helper equations
9.1. Overview
The equations below are used in multiple places by the Vorbis codec specification. Rather
than cluttering up the main specification documents, they are defined here and referenced
where appropriate.
9.2. Functions
9.2.1. ilog
The ilog(x) function returns the position number (1 through n) of the highest set bit in
the twos complement integer value [x]. Values of [x] less than zero are defined to return
zero.
1
2
1) [return\_value] = 0;
2) if ( [x] is greater than zero ) {
3) increment [return\_value];
4) logical shift [x] one bit to the right, padding the MSb with zero
5) repeat at step 2)
4
5
6
7
8
9
10
6) done
Examples:
ilog(0) = 0;
ilog(1) = 1;
ilog(2) = 2;
ilog(3) = 2;
ilog(4) = 3;
ilog(7) = 3;
ilog(negative number) = 0;
9.2.2. float32 unpack
float32 unpack(x) is intended to translate the packed binary representation of a Vorbis
codebook float value into the representation used by the decoder for floating point numbers.
63
For purposes of this example, we will unpack a Vorbis float32 into a host-native floating
point number.
1
2
3
4
5
1)
2)
3)
4)
5)
[mantissa] = [x] bitwise AND 0x1fffff (unsigned result)

[sign] = [x] bitwise AND 0x80000000 (unsigned result)
[exponent] = ( [x] bitwise AND 0x7fe00000) shifted right 21 bits (unsigned result)
if ( [sign] is nonzero ) then negate [mantissa]
return [mantissa] * ( 2 ^ ( [exponent] - 788 ) )
9.2.3. lookup1 values

lookup1 values(codebook entries,codebook dimensions) is used to compute the correct
length of the value index for a codebook VQ lookup table of lookup type 1. The values on
this list are permuted to construct the VQ vector lookup table of size [codebook_entries].
The return value for this function is defined to be the greatest integer value for which
[return_value] to the power of [codebook_dimensions] is less than or equal to [codebook_
entries].
9.2.4. low neighbor
low neighbor(v,x) finds the position n in vector [v] of the greatest value scalar element
for which n is less than [x] and vector [v] element n is less than vector [v] element [x].
9.2.5. high neighbor
high neighbor(v,x) finds the position n in vector [v] of the lowest value scalar element
for which n is less than [x] and vector [v] element n is greater than vector [v] element
[x].
9.2.6. render point
render point(x0,y0,x1,y1,X) is used to find the Y value at point X along the line specified
by x0, x1, y0 and y1. This function uses an integer algorithm to solve for the point directly
without calculating intervening values along the line.
1
2
3
4
5
6
1)
2)
3)
4)
5)
6)
[dy] = [y1] - [y0]

[adx] = [x1] - [x0]
[ady] = absolute value of [dy]
[err] = [ady] * ([X] - [x0])
[off] = [err] / [adx] using integer division
if ( [dy] is less than zero ) {
7
8
7) [Y] = [y0] - [off]
9
10
} else {
11
64
8) [Y] = [y0] + [off]
12
13
14
15
16
9) done
9.2.7. render line

Floor decode type one uses the integer line drawing algorithm of render line(x0, y0, x1,
y1, v) to construct an integer floor curve for contiguous piecewise line segments. Note that
it has not been relevant elsewhere, but here we must define integer division as rounding
division of both positive and negative numbers toward zero.
1
2
3
4
5
6
7
1)
[dy] = [y1] - [y0]
2) [adx] = [x1] - [x0]
3) [ady] = absolute value of [dy]
4) [base] = [dy] / [adx] using integer division
5)
[x] = [x0]
6)
[y] = [y0]
7) [err] = 0
8
9
8) if ( [dy] is less than 0 ) {
10
9) [sy] = [base] - 1
11
12
13
} else {
14
10) [sy] = [base] + 1
15
16
17
18
19
20
11) [ady] = [ady] - (absolute value of [base]) * [adx]

12) vector [v] element [x] = [y]
21
22
13) iterate [x] over the range [x0]+1 ... [x1]-1 {
23
14) [err] = [err] + [ady];

15) if ( [err] >= [adx] ) {
24
25
26
16) [err] = [err] - [adx]

17)
[y] = [y] + [sy]
27
28
29
} else {
30
31
18) [y] = [y] + [base]
32
33
34
35
19) vector [v] element [x] = [y]
36
37
38
65
10. Tables
10.1. floor1 inverse dB table
The vector [floor1_inverse_dB_table] is a 256 element static lookup table consiting of
the following values (read left to right then top to bottom):
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
1.0649863e-07,
1.3699951e-07,
1.7623575e-07,
2.2670913e-07,
2.9163793e-07,
3.7516214e-07,
4.8260743e-07,
6.2082472e-07,
7.9862701e-07,
1.0273513e-06,
1.3215816e-06,
1.7000785e-06,
2.1869758e-06,
2.8133190e-06,
3.6190449e-06,
4.6555282e-06,
5.9888572e-06,
7.7040476e-06,
9.9104632e-06,
1.2748789e-05,
1.6400004e-05,
2.1096914e-05,
2.7139006e-05,
3.4911534e-05,
4.4910090e-05,
5.7772202e-05,
7.4317983e-05,
9.5602426e-05,
0.00012298267,
0.00015820453,
0.00020351382,
0.00026179955,
0.00033677814,
0.00043323036,
0.00055730621,
0.00071691700,
0.00092223983,
0.0011863665,
0.0015261382,
0.0019632195,
0.0025254795,
0.0032487691,
0.0041792066,
0.0053761186,
0.0069158225,
0.0088964928,
0.011444421,
0.014722068,
0.018938423,
0.024362330,
0.031339626,
0.040315199,
0.051861348,
0.066714279,
1.1341951e-07,
1.4590251e-07,
1.8768855e-07,
2.4144197e-07,
3.1059021e-07,
3.9954229e-07,
5.1396998e-07,
6.6116941e-07,
8.5052630e-07,
1.0941144e-06,
1.4074654e-06,
1.8105592e-06,
2.3290978e-06,
2.9961443e-06,
3.8542308e-06,
4.9580707e-06,
6.3780469e-06,
8.2047000e-06,
1.0554501e-05,
1.3577278e-05,
1.7465768e-05,
2.2467911e-05,
2.8902651e-05,
3.7180282e-05,
4.7828601e-05,
6.1526565e-05,
7.9147585e-05,
0.00010181521,
0.00013097477,
0.00016848555,
0.00021673929,
0.00027881276,
0.00035866388,
0.00046138411,
0.00059352311,
0.00076350630,
0.00098217216,
0.0012634633,
0.0016253153,
0.0020908006,
0.0026895994,
0.0034598925,
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0.0057254891,
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0.009474637,
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0.015678791,
0.020169149,
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0.033376252,
0.042935108,
0.055231591,
0.071049749,
1.2079015e-07,
1.5538408e-07,
1.9988561e-07,
2.5713223e-07,
3.3077411e-07,
4.2550680e-07,
5.4737065e-07,
7.0413592e-07,
9.0579828e-07,
1.1652161e-06,
1.4989305e-06,
1.9282195e-06,
2.4804557e-06,
3.1908506e-06,
4.1047004e-06,
5.2802740e-06,
6.7925283e-06,
8.7378876e-06,
1.1240392e-05,
1.4459606e-05,
1.8600792e-05,
2.3928002e-05,
3.0780908e-05,
3.9596466e-05,
5.0936773e-05,
6.5524908e-05,
8.4291040e-05,
0.00010843174,
0.00013948625,
0.00017943469,
0.00023082423,
0.00029693158,
0.00038197188,
0.00049136745,
0.00063209358,
0.00081312324,
0.0010459992,
0.0013455702,
0.0017309374,
0.0022266726,
0.0028643847,
0.0036847358,
0.0047400328,
0.0060975636,
0.0078438871,
0.010090352,
0.012980198,
0.016697687,
0.021479854,
0.027631618,
0.035545228,
0.045725273,
0.058820850,
0.075666962,
1.2863978e-07,
1.6548181e-07,
2.1287530e-07,
2.7384213e-07,
3.5226968e-07,
4.5315863e-07,
5.8294187e-07,
7.4989464e-07,
9.6466216e-07,
1.2409384e-06,
1.5963394e-06,
2.0535261e-06,
2.6416497e-06,
3.3982101e-06,
4.3714470e-06,
5.6234160e-06,
7.2339451e-06,
9.3057248e-06,
1.1970856e-05,
1.5399272e-05,
1.9809576e-05,
2.5482978e-05,
3.2781225e-05,
4.2169667e-05,
5.4246931e-05,
6.9783085e-05,
8.9768747e-05,
0.00011547824,
0.00014855085,
0.00019109536,
0.00024582449,
0.00031622787,
0.00040679456,
0.00052329927,
0.00067317058,
0.00086596457,
0.0011139742,
0.0014330129,
0.0018434235,
0.0023713743,
0.0030505286,
0.0039241906,
0.0050480668,
0.0064938176,
0.0083536271,
0.010746080,
0.013823725,
0.017782797,
0.022875735,
0.029427276,
0.037855157,
0.048696758,
0.062643361,
0.080584227,
66
55
56
57
58
59
60
61
62
63
64
0.085821044,
0.11039993,
0.14201813,
0.18269168,
0.23501402,
0.30232132,
0.38890521,
0.50028648,
0.64356699,
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0.091398179,
0.11757434,
0.15124727,
0.19456402,
0.25028656,
0.32196786,
0.41417847,
0.53279791,
0.68538959,
0.88168307,
0.097337747,
0.12521498,
0.16107617,
0.20720788,
0.26655159,
0.34289114,
0.44109412,
0.56742212,
0.72993007,
0.9389798,
0.10366330,
0.13335215,
0.17154380,
0.22067342,
0.28387361,
0.36517414,
0.46975890,
0.60429640,
0.77736504,
1.
67
A. Embedding Vorbis into an Ogg stream

A.1. Overview
This document describes using Ogg logical and physical transport streams to encapsulate
Vorbis compressed audio packet data into file form.
The Section 1, Introduction and Description provides an overview of the construction of
Vorbis audio packets.
The Ogg bitstream overview and Ogg logical bitstream and framing spec provide detailed
descriptions of Ogg transport streams. This specification document assumes a working
knowledge of the concepts covered in these named backround documents. Please read
them first.
A.1.1. Restrictions
The Ogg/Vorbis I specification currently dictates that Ogg/Vorbis streams use Ogg transport streams in degenerate, unmultiplexed form only. That is:
A meta-headerless Ogg file encapsulates the Vorbis I packets
The Ogg stream may be chained, i.e., contain multiple, contigous logical streams
(links).
The Ogg stream must be unmultiplexed (only one stream, a Vorbis audio stream,
per link)
This is not to say that it is not currently possible to multiplex Vorbis with other media
types into a multi-stream Ogg file. At the time this document was written, Ogg was
becoming a popular container for low-bitrate movies consisting of DivX video and Vorbis
audio. However, a Vorbis I audio file is taken to imply Vorbis audio existing alone within a
degenerate Ogg stream. A compliant Vorbis audio player is not required to implement Ogg
support beyond the specific support of Vorbis within a degenrate Ogg stream (naturally,
application authors are encouraged to support full multiplexed Ogg handling).
A.1.2. MIME type
The MIME type of Ogg files depend on the context. Specifically, complex multimedia and
applications should use application/ogg, while visual media should use video/ogg, and
audio audio/ogg. Vorbis data encapsulated in Ogg may appear in any of those types.
RTP encapsulated Vorbis should use audio/vorbis + audio/vorbis-config.
68
A.2. Encapsulation
Ogg encapsulation of a Vorbis packet stream is straightforward.
The first Vorbis packet (the identification header), which uniquely identifies a stream
as Vorbis audio, is placed alone in the first page of the logical Ogg stream. This
results in a first Ogg page of exactly 58 bytes at the very beginning of the logical
stream.
This first page is marked beginning of stream in the page flags.
The second and third vorbis packets (comment and setup headers) may span one or
more pages beginning on the second page of the logical stream. However many pages
they span, the third header packet finishes the page on which it ends. The next (first
audio) packet must begin on a fresh page.
The granule position of these first pages containing only headers is zero.
The first audio packet of the logical stream begins a fresh Ogg page.
Packets are placed into ogg pages in order until the end of stream.
The last page is marked end of stream in the page flags.
Vorbis packets may span page boundaries.
The granule position of pages containing Vorbis audio is in units of PCM audio
samples (per channel; a stereo streams granule position does not increment at twice
the speed of a mono stream).
The granule position of a page represents the end PCM sample position of the last
packet completed on that page. The last PCM sample is the last complete sample
returned by decode, not an internal sample awaiting lapping with a subsequent block.
A page that is entirely spanned by a single packet (that completes on a subsequent
page) has no granule position, and the granule position is set to -1.
Note that the last decoded (fully lapped) PCM sample from a packet is not necessarily
the middle sample from that block. If, eg, the current Vorbis packet encodes a
long block and the next Vorbis packet encodes a short block, the last decodable
sample from the current packet be at position (3*long block length/4) - (short block
length/4).
The granule (PCM) position of the first page need not indicate that the stream
started at position zero. Although the granule position belongs to the last completed
packet on the page and a valid granule position must be positive, by inference it may
indicate that the PCM position of the beginning of audio is positive or negative.
A positive starting value simply indicates that this stream begins at some positive time offset, potentially within a larger program. This is a common case
69
when connecting to the middle of broadcast stream.

A negative value indicates that output samples preceeding time zero should be
discarded during decoding; this technique is used to allow sample-granularity
editing of the stream start time of already-encoded Vorbis streams. The number
of samples to be discarded must not exceed the overlap-add span of the first two
audio packets.
In both of these cases in which the initial audio PCM starting offset is nonzero, the
second finished audio packet must flush the page on which it appears and the third
packet begin a fresh page. This allows the decoder to always be able to perform
PCM position adjustments before needing to return any PCM data from synthesis,
resulting in correct positioning information without any aditional seeking logic.
Note: Failure to do so should, at worst, cause a decoder implementation to return
incorrect positioning information for seeking operations at the very beginning of the
stream.
A granule position on the final page in a stream that indicates less audio data than
the final packet would normally return is used to end the stream on other than even
frame boundaries. The difference between the actual available data returned and the
declared amount indicates how many trailing samples to discard from the decoding
process.
70
B. Vorbis encapsulation in RTP

Please consult RFC 5215 RTP Payload Format for Vorbis Encoded Audio for description
of how to embed Vorbis audio in an RTP stream.
71
Colophon
Ogg is a Xiph.Org Foundation effort to protect essential tenets of Internet multimedia from
corporate hostage-taking; Open Source is the nets greatest tool to keep everyone honest.
See About the Xiph.Org Foundation for details.
Ogg Vorbis is the first Ogg audio CODEC. Anyone may freely use and distribute the Ogg
and Vorbis specification, whether in a private, public or corporate capacity. However,
the Xiph.Org Foundation and the Ogg project (xiph.org) reserve the right to set the Ogg
Vorbis specification and certify specification compliance.
Xiph.Orgs Vorbis software CODEC implementation is distributed under a BSD-like license. This does not restrict third parties from distributing independent implementations
of Vorbis software under other licenses.
Ogg, Vorbis, Xiph.Org Foundation and their logos are trademarks (tm) of the Xiph.Org
Foundation. These pages are copyright (C) 1994-2007 Xiph.Org Foundation. All rights
reserved.
This document is set using LATEX.
72
References
[1] T. Sporer, K. Brandenburg and B. Edler, The use of multirate filter banks for coding
of high quality digital audio, http://www.iocon.com/resource/docs/ps/eusipco_
corrected.ps.
73

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Vorbis I specification

2.1.8. end-of-packet alignment . . . . . . . . . . . . . . . . . . . . . . . .

4. Codec Setup and Packet Decode

7. Floor type 1 setup and decode

8. Residue setup and decode

A. Embedding Vorbis into an Ogg stream

B. Vorbis encapsulation in RTP

1. Introduction and Description

software/interface designs, such as Windows ACM codec framework). However, we find

1.2. Decoder Configuration

Figure 1: decoder pipeline configuration

1.2.1. Global Config

1.3. High-level Decode Process

6. inverse channel coupling of residue vectors

Figure 2: overlap of two equal-sized windows

Figure 3: overlap of a long and a short window

bytestream length == 1 byte

Continue by coding the 3 bit integer value -1 [b111]:

bytestream length == 1 byte

Continue by coding the 7 bit integer value 17 [b0010001]:

bytestream length == 2 bytes

<bytestream length == 4 bytes

2.1.7. decoding example

bytestream length == 4 bytes

3. Probability Model and Codebooks

3.2. Packed codebook format

16 bit [codebook_dimensions] and 24 bit [codebook_entries] fields:

Next is the [ordered] bit flag:

Each entry, numbering a total of [codebook_entries], is assigned a codeword length.

X 1 ] [sparse] flag (1 bit)

2) [flag] = read one bit;

4) [length] = read a five bit unsigned integer;

6) this entry is unused.

} else the sparse flag is not set {

7) [length] = read a five bit unsigned integer;

1) [codebook_lookup_type] = read four bits as an unsigned integer

Codebook decode precedes according to [codebook_lookup_type]:

[codebook_minimum_value] = float32_unpack( read 32 bits as an unsigned integer)

5) [codebook_lookup_values] = lookup1_values([codebook_entries], [codebook_dimensions] )

6) [codebook_lookup_values] = [codebook_entries] * [codebook_dimensions]

7) read a total of [codebook_lookup_values] unsigned integers of [codebook_value_bits] each;

A [codebook_lookup_type] of greater than two is reserved and indicates a stream

possible. Assume the following codeword length list:

Figure 4: huffman tree illustration

Figure 5: underspecified huffman tree illustration

the [codebook\_multiplicands] array

4) [multiplicand_offset] = ( [lookup_offset] divided by [index_divisor] using integer

5) vector [value_vector] element [i] =

6) if ( [codebook_sequence_p] is set ) then set [last] = vector [value_vector] element [i]

7) [index_divisor] = [index_divisor] * [codebook_lookup_values]

8) vector calculation completed.

4) vector [value_vector] element [i] =

5) if ( [codebook_sequence_p] is set ) then set [last] = vector [value_vector] element [i]

7) vector calculation completed.

3.3. Use of the codebook abstraction

4. Codec Setup and Packet Decode

4.2. Header decode and decode setup

1) [packet_type] : 8 bit value

[vorbis_version] = read 32 bits as unsigned integer