Ogg Vorbis encoding format documentation

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wait As of writing, not all the below document -links are live. They will be populated as we complete the documents.

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Documents

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Programming with libvorbis

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Description

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Ogg Vorbis is a general purpose compressed audio format -for high quality (44.1-48.0kHz, 16+ bit, polyphonic) audio and music -at moderate fixed and variable bitrates (40-80 kb/s/channel). This -places Vorbis in the same class as audio representations including -MPEG-1 audio layer 3, MPEG-4 audio (AAC and TwinVQ), and PAC.

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Vorbis is the first of a planned family of Ogg multimedia coding -formats being developed as part of the Xiph.Org Foundation's Ogg multimedia -project. See http://www.xiph.org/ -for more information.

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Vorbis technical documents

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A Vorbis encoder takes in overlapping (but contiguous) short-time -segments of audio data. The encoder analyzes the content of the audio -to determine an optimal compact representation; this phase of encoding -is known as analysis. For each short-time block of sound, -the encoder then packs an efficient representation of the signal, as -determined by analysis, into a raw packet much smaller than the size -required by the original signal; this phase is coding. -Lastly, in a streaming environment, the raw packets are then -structured into a continuous stream of octets; this last phase is -streaming. Note that the stream of octets is referred to both -as a 'byte-' and 'bit-'stream; the latter usage is acceptible as the -stream of octets is a physical representation of a true logical -bit-by-bit stream.

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A Vorbis decoder performs a mirror image process of extracting the -original sequence of raw packets from an Ogg stream (stream -decomposition), reconstructing the signal representation from the -raw data in the packet (decoding) and them reconstituting an -audio signal from the decoded representation (synthesis).

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The Programming with libvorbis -documents discuss use of the reference Vorbis codec library -(libvorbis) produced by the Xiph.Org Foundation.

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The data representations and algorithms necessary at each step to -encode and decode Ogg Vorbis bitstreams are described by the below -documents in sufficient detail to construct a complete Vorbis codec. -Note that at the time of writing, Vorbis is still in a 'Request For -Comments' stage of development; despite being in advanced stages of -development, input from the multimedia community is welcome.

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Vorbis analysis and synthesis

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Analysis begins by seperating an input audio stream into individual, -overlapping short-time segments of audio data. These segments are -then transformed into an alternate representation, seeking to -represent the original signal in a more efficient form that codes into -a smaller number of bytes. The analysis and transformation stage is -the most complex element of producing a Vorbis bitstream.

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The corresponding synthesis step in the decoder is simpler; there is -no analysis to perform, merely a mechanical, deterministic -reconstruction of the original audio data from the transform-domain -representation.

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Vorbis packet structure: -Describes the basic analysis components necessary to produce Vorbis -packets and the structure of the packet itself.
Temporal envelope shaping and blocksize: -Use of temporal envelope shaping and variable blocksize to minimize -time-domain energy leakage during wide dynamic range and spectral energy -swings. Also discusses time-related principles of psychoacoustics.
Time domain segmentation and MDCT transform: -Division of time domain data into individual overlapped, windowed -short-time vectors and transformation using the MDCT
The resolution floor: Use of frequency -doamin psychoacoustics, and the MDCT-domain noise, masking and resolution -floors
MDCT-domain fine structure: Production, -quantization and massaging of MDCT-spectrum fine structure

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Vorbis coding and decoding

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Coding and decoding converts the transform-domain representation of -the original audio produced by analysis to and from a bitwise packed -raw data packet. Coding and decoding consist of two logically -orthogonal concepts, back-end coding and bitpacking.

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Back-end coding uses a probability model to represent the raw numbers -of the audio representation in as few physical bits as possible; -familiar examples of back-end coding include Huffman coding and Vector -Quantization.

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Bitpacking arranges the variable sized words of the back-end -coding into a vector of octets without wasting space. The octets -produced by coding a single short-time audio segment is one raw Vorbis -packet.

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The Vorbis probability model
The Vorbis bitpacker: Arrangement of -variable bit-length words into an octet-aligned packet.

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Vorbis streaming and stream decomposition

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Vorbis packets contain the raw, bitwise-compressed representation of a -snippet of audio. These packets contain no structure and cannot be -strung together directly into a stream; for streamed transmission and -storage, Vorbis packets are encoded into an Ogg bitstream.

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Ogg bitstream overview: High-level -description of Ogg logical bitstreams, how logical bitstreams -(of mixed media types) can be combined into physical bitstreams, and -restrictions on logical-to-physical mapping. Note that this document is -not specific only to Ogg Vorbis.
Ogg logical bitstream and framing -spec: Low level, complete specification of Ogg logical -bitstream pages. Note that this document is not specific only to Ogg -Vorbis.
Vorbis bitstream mapping: -Specifically describes mapping Vorbis data into an -Ogg physical bitstream.

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