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- % -*- mode: latex; TeX-master: "Vorbis_I_spec"; -*-
- %!TEX root = Vorbis_I_spec.tex
- % $Id$
- \section{Introduction and Description} \label{vorbis:spec:intro}
- \subsection{Overview}
- This document provides a high level description of the Vorbis codec's
- construction. A bit-by-bit specification appears beginning in
- \xref{vorbis:spec:codec}.
- The later sections assume a high-level
- understanding of the Vorbis decode process, which is
- provided here.
- \subsubsection{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).
- \subsubsection{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.
- \subsubsection{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 \xref{vorbis:over:ogg}.
- \subsubsection{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.
- \begin{note}
- Vorbis \emph{can} initiate decode at any arbitrary packet within a
- bitstream so long as the codec has been initialized/setup with the
- setup headers.
- \end{note}
- 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.Org's 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.
- \subsubsection{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.
- \subsubsection{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.
- \subsection{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:
- \begin{center}
- \includegraphics[width=\textwidth]{components}
- \captionof{figure}{decoder pipeline configuration}
- \end{center}
- \subsubsection{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.
- \subsubsection{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.
- \subsubsection{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
- submap's 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.
- \subsubsection{Floor}
- Vorbis encodes a spectral 'floor' vector for each PCM channel. This
- vector is a low-resolution 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
- coupled-stereo 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.org's 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.
- \subsubsection{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.
- \subsubsection{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).
- \subsection{High-level Decode Process}
- \subsubsection{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.
- \paragraph{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.
- \paragraph{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 \xref{vorbis:spec:comment}.
- \paragraph{Setup Header}
- The setup header includes extensive CODEC setup information as well as
- the complete VQ and Huffman codebooks needed for decode.
- \subsubsection{Decode Procedure}
- The decoding and synthesis procedure for all audio packets is
- fundamentally the same.
- \begin{enumerate}
- \item decode packet type flag
- \item decode mode number
- \item decode window shape (long windows only)
- \item decode floor
- \item decode residue into residue vectors
- \item inverse channel coupling of residue vectors
- \item generate floor curve from decoded floor data
- \item compute dot product of floor and residue, producing audio spectrum vector
- \item inverse monolithic transform of audio spectrum vector, always an MDCT in Vorbis I
- \item overlap/add left-hand output of transform with right-hand output of previous frame
- \item store right hand-data from transform of current frame for future lapping
- \item if not first frame, return results of overlap/add as audio result of current frame
- \end{enumerate}
- 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 \emph{entirely mathematically equivalent} to the
- specification, it need not be a literal semantic implementation.
- \paragraph{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; \emph{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}.
- \paragraph{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.
- \paragraph{Window shape decode (long windows only)} \label{vorbis:spec:window}
- 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:
- \begin{center}
- \includegraphics[width=\textwidth]{window1}
- \captionof{figure}{overlap of two equal-sized windows}
- \end{center}
- And slightly more complex in the case of overlapping unequal sized
- windows:
- \begin{center}
- \includegraphics[width=\textwidth]{window2}
- \captionof{figure}{overlap of a long and a short window}
- \end{center}
- 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 post- window 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 \cite{Sporer/Brandenburg/Edler}. Vorbis windows
- all use the slope function
- \[ y = \sin(.5*\pi \, \sin^2((x+.5)/n*\pi)) . \]
- \paragraph{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.
- \paragraph{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.
- \paragraph{inverse channel coupling}
- A detailed discussion of stereo in the Vorbis codec can be found in
- the document \href{stereo.html}{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.
- \paragraph{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.
- \paragraph{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.
- % TODO/FIXME: The following two paragraphs have identical twins
- % in section 4 (under "dot product")
- 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 a 16 bit PCM device. For the
- residue vector to represent full scale if the floor is nailed to
- $-140$dB, it must be able to span 0 to $+140$dB. For the residue vector
- to reach full scale if the floor is nailed at 0dB, it must be able to
- represent $-140$dB to $+0$dB. Thus, in order to handle full range
- dynamics, a residue vector may span $-140$dB to $+140$dB 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.
- \paragraph{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 \cite{Sporer/Brandenburg/Edler}.
- 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.
- \paragraph{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.
- \paragraph{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.
- \paragraph{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:
- \begin{Verbatim}[commandchars=\\\{\}]
- window_blocksize(previous_window)/4+window_blocksize(current_window)/4
- \end{Verbatim}
- 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).
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