As of writing, not all the below document
links are live. They will be populated as we complete the documents.
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.
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.
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.
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).
The Programming with libvorbis documents discuss use of the reference Vorbis codec library (libvorbis) produced by the Xiph.Org Foundation.
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.
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.
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.
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.
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.
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.
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.