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