From cebf8a2c2f1d7395341e21ab8f04ee9795f02622 Mon Sep 17 00:00:00 2001 From: Ralph Giles Date: Mon, 14 Oct 2002 23:47:54 +0000 Subject: [PATCH] fix crossreferences. svn path=/trunk/vorbis/; revision=4003 --- doc/xml/01-introduction.xml | 29 +++++++++++++++-------------- doc/xml/03-codebook.xml | 33 ++++++++++++--------------------- doc/xml/04-codec.xml | 10 +++++----- 3 files changed, 32 insertions(+), 40 deletions(-) diff --git a/doc/xml/01-introduction.xml b/doc/xml/01-introduction.xml index 691256a..fd13c51 100644 --- a/doc/xml/01-introduction.xml +++ b/doc/xml/01-introduction.xml @@ -1,7 +1,7 @@
- $Id: 01-introduction.xml,v 1.2 2002/10/13 15:18:46 giles Exp $ + $Id: 01-introduction.xml,v 1.3 2002/10/14 23:47:54 giles Exp $ Last update to this document: July 18, 2002 @@ -12,11 +12,10 @@ This document provides a high level description of the Vorbis codec's -construction. A bit-by-bit specification appears beginning in the packet specification and reference -document. The other reference documents assumes a high-level -understanding of the Vorbis decode process, which is provided in this -document. +construction. A bit-by-bit specification appears beginning in . The other reference documents assumes +a high-level understanding of the Vorbis decode process, which is +provided in this document.
Application @@ -469,10 +468,11 @@ decode. A description of valid window functions for use with an inverse MDCT -can be found in the paper _The -use of multirate filter banks for coding of high quality digital -audio_, by T. Sporer, K. Brandenburg and B. Edler. Vorbis windows +can be found in the paper + + +The use of multirate filter banks for coding of high quality digital +audio, by T. Sporer, K. Brandenburg and B. Edler. Vorbis windows all use the slope function y=sin(2PI*sin^2(x/n)) @@ -590,10 +590,11 @@ representation. 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 the paper _The -use of multirate filter banks for coding of high quality digital -audio_, by T. Sporer, K. Brandenburg and B. Edler. +description of the MDCT is available in the paper + +The use of multirate filter banks for coding of high quality digital +audio, by T. Sporer, K. Brandenburg and B. Edler. Note that the PCM produced directly from the MDCT is not yet finished diff --git a/doc/xml/03-codebook.xml b/doc/xml/03-codebook.xml index 8cbdc59..43b3da0 100644 --- a/doc/xml/03-codebook.xml +++ b/doc/xml/03-codebook.xml @@ -1,7 +1,7 @@
- $Id: 03-codebook.xml,v 1.1 2002/10/12 20:37:11 giles Exp $ + $Id: 03-codebook.xml,v 1.2 2002/10/14 23:47:54 giles Exp $ Last update to this document: August 8, 2002 @@ -23,12 +23,10 @@ decoded output corresponding to a given compressed codeword.
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 arbitrary-width values read from the stream -according to the Vorbis bitpacking -convention. +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 arbitrary-width values read from the +stream according to .
@@ -39,9 +37,8 @@ convention. For purposes of the below examples, we assume that the storage system's native byte width is eight bits. This is not universally -true; see the Vorbis bitpacking -convention document for discussion relating to non-eight-bit -bytes. +true; see for discussion +relating to non-eight-bit bytes.
codebook decode @@ -137,8 +134,7 @@ byte 8: [ X 1 ] [sparse] flag (1 bit) 1) [current_entry] = 0; 2) [current_length] = read a five bit unsigned integer and add 1; - 3) [number] = read ilog([codebook_entries] - + 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, @@ -189,19 +185,14 @@ vector explicitly, rather than building vectors from a smaller list of possible scalar values. Lookup decode proceeds as follows: - 1) [codebook_minimum_value] = -float32_unpack( read 32 -bits as an unsigned integer) - 2) [codebook_delta_value] = float32_unpack( read 32 bits as -an unsigned integer) + 1) [codebook_minimum_value] = float32_unpack( read 32 bits as an unsigned integer) + 2) [codebook_delta_value] = float32_unpack( read 32 bits as an unsigned integer) 3) [codebook_value_bits] = read 4 bits as an unsigned integer and add 1 4) [codebook_sequence_p] = read 1 bit as a boolean flag if ( [codebook_lookup_type] is 1 ) { - 5) [codebook_lookup_values] = -lookup1_values( -[codebook_entries], [codebook_dimensions] ) + 5) [codebook_lookup_values] = lookup1_values([codebook_entries], [codebook_dimensions] ) } else { @@ -340,7 +331,7 @@ the [codebook_multiplicands] array -Decoding [unpacking] a specific vector in the vector lookup table +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. diff --git a/doc/xml/04-codec.xml b/doc/xml/04-codec.xml index 8f622d2..d77447e 100644 --- a/doc/xml/04-codec.xml +++ b/doc/xml/04-codec.xml @@ -1,7 +1,7 @@
- $Id: 04-codec.xml,v 1.1 2002/10/12 20:37:11 giles Exp $ + $Id: 04-codec.xml,v 1.2 2002/10/14 23:47:54 giles Exp $ Last update to this document: September 20, 2002 @@ -619,10 +619,10 @@ representation. 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 the paper -_The +The use of multirate filter banks for coding of high quality digital -audio_, by T. Sporer, K. Brandenburg and B. Edler. The window +audio, by T. Sporer, K. Brandenburg and B. Edler. The window function used for the MDCT is the window determined earlier.
@@ -704,7 +704,7 @@ center, front right, rear left, rear right, LFE Applications using Vorbis for dedicated purposes may define channel mapping as seen fit. Future channel mappings (such as three and four -channel Ambisonics) will +channel Ambisonics) will make use of channel mappings other than mapping 0.
-- 2.7.4