comparison src/libvorbis-1.3.3/doc/01-introduction.tex @ 1:05aa0afa9217

Bring in flac, ogg, vorbis

+% -*- 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).