diff --git a/doc/draft-ietf-codec-opus.xml b/doc/draft-ietf-codec-opus.xml index 9e3a281d..08085fff 100644 --- a/doc/draft-ietf-codec-opus.xml +++ b/doc/draft-ietf-codec-opus.xml @@ -83,7 +83,7 @@ It is composed of a linear prediction (LP)-based layer and a Modified Discrete Cosine Transform (MDCT)-based layer. The main idea behind using two layers is that in speech, linear prediction - techniques (such as CELP) code low frequencies more efficiently than transform + techniques (such as Code-Excited Linear Prediction, or CELP) code low frequencies more efficiently than transform (e.g., MDCT) domain techniques, while the situation is reversed for music and higher speech frequencies. Thus a codec with both layers available can operate over a wider range than @@ -150,7 +150,8 @@ E.g., the text will explicitly indicate any shifts required after a Expressions, where included in the text, follow C operator rules and precedence, with the exception that the syntax "x**y" indicates x raised to - the power y. + the power y. Throughout this document, the term "byte" is defined to include 8 bits, + i.e. an octet. The text also makes use of the following functions: @@ -221,6 +222,12 @@ Examples: +
+ +Largest integer z such that z <= x. + +
+ @@ -279,7 +286,7 @@ It supports NB, MB, or WB audio and frame sizes from 10 ms to 60 ms, and requires an additional 5 ms look-ahead for noise shaping estimation. A small additional delay (up to 1.5 ms) may be required for sampling rate conversion. -Like Vorbis and many other modern codecs, SILK is inherently designed for +Like Vorbis and many other modern codecs, SILK is inherently designed for variable-bitrate (VBR) coding, though the encoder can also produce constant-bitrate (CBR) streams. The version of SILK used in Opus is substantially modified from, and not @@ -477,7 +484,8 @@ is required. There are two main reasons to operate in CBR mode: When low-latency transmission is required over a relatively slow connection, then constrained VBR can also be used. This uses VBR in a way that simulates a -"bit reservoir" and is equivalent to what MP3 and AAC call CBR (i.e. not true +"bit reservoir" and is equivalent to what MP3 (MPEG 1, Layer 3) and +AAC (Advanced Audio Coding) call CBR (i.e. not true CBR due to the bit reservoir). @@ -507,7 +515,8 @@ A single packet may contain multiple audio frames, so long as they share a This section describes the possible combinations of these parameters and the internal framing used to pack multiple frames into a single packet. This framing is not self-delimiting. -Instead, it assumes that a higher layer (such as UDP or RTP or Ogg or Matroska) +Instead, it assumes that a higher layer (such as UDP or RTP +or Ogg or Matroska ) will communicate the length, in bytes, of the packet, and it uses this information to reduce the framing overhead in the packet itself. A decoder implementation MUST support the framing described in this section. @@ -1000,7 +1009,8 @@ stream | Range |---+ +---------+ +------------+ /---\ Audio
-Opus uses an entropy coder based on , +Opus uses an entropy coder based on range coding +, which is itself a rediscovery of the FIFO arithmetic code introduced by . It is very similar to arithmetic encoding, except that encoding is done with digits in any base instead of with bits, @@ -6148,7 +6158,7 @@ The procedure in does this in a way that The function ec_enc_uint() (entenc.c) encodes one of ft equiprobable symbols in the range 0 to (ft - 1), inclusive, each with a frequency of 1, where ft may be as large as (2**32 - 1). -Like the decoder (see ), it splits it splits up the +Like the decoder (see ), it splits up the value into a range coded symbol representing up to 8 of the high bits, and, if necessary, raw bits representing the remainder of the value. @@ -7489,6 +7499,9 @@ name of work, or endorsement information. + + + SILK Speech Codec @@ -7590,7 +7603,7 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect - + Range encoding: An algorithm for removing redundancy from a digitised message @@ -7654,6 +7667,20 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect + + +Vorbis website + + + + + + +Matroska website + + + + Opus Testvectors (webside) @@ -7668,6 +7695,13 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect + + +Range Coding +Wikipedia + + + Hadamard Transform