First set of changes for Gen-art review
This commit is contained in:
parent
516c980585
commit
e8c437c432
1 changed files with 42 additions and 8 deletions
|
@ -83,7 +83,7 @@ It is composed of a linear
|
|||
prediction (LP)-based <xref target="LPC"/> layer and a Modified Discrete Cosine Transform
|
||||
(MDCT)-based <xref target="MDCT"/> 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
|
|||
<t>
|
||||
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:
|
||||
</t>
|
||||
|
||||
|
@ -221,6 +222,12 @@ Examples:
|
|||
</t>
|
||||
</section>
|
||||
|
||||
<section anchor="floor" toc="exclude" title="floor(x)">
|
||||
<t>
|
||||
Largest integer z such that z <= x.
|
||||
</t>
|
||||
</section>
|
||||
|
||||
</section>
|
||||
|
||||
</section>
|
||||
|
@ -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 <xref target='Vorbis-website'/> 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).
|
||||
</t>
|
||||
</section>
|
||||
|
@ -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 <xref target='RFC3550'/>
|
||||
or Ogg <xref target='RFC3533'/> or Matroska <xref target='Matroska-website'/>)
|
||||
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
|
|||
|
||||
<section anchor="range-decoder" title="Range Decoder">
|
||||
<t>
|
||||
Opus uses an entropy coder based on <xref target="range-coding"></xref>,
|
||||
Opus uses an entropy coder based on range coding <xref target="range-coding"></xref>
|
||||
<xref target="Nigel79"></xref>,
|
||||
which is itself a rediscovery of the FIFO arithmetic code introduced by <xref target="coding-thesis"></xref>.
|
||||
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 <xref target="encoder-finalizing"/> 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 <xref target="ec_dec_uint"/>), it splits it splits up the
|
||||
Like the decoder (see <xref target="ec_dec_uint"/>), 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.
|
||||
</t>
|
||||
|
@ -7489,6 +7499,9 @@ name of work, or endorsement information.</t>
|
|||
<format type='TXT' target='http://tools.ietf.org/rfc/rfc6366.txt' />
|
||||
</reference>
|
||||
|
||||
<?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3550.xml"?>
|
||||
<?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.3533.xml"?>
|
||||
|
||||
<reference anchor='SILK' target='http://developer.skype.com/silk'>
|
||||
<front>
|
||||
<title>SILK Speech Codec</title>
|
||||
|
@ -7590,7 +7603,7 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
|
|||
<format type='TXT' octets='110393' target='ftp://ftp.isi.edu/in-notes/rfc3552.txt' />
|
||||
</reference>
|
||||
|
||||
<reference anchor="range-coding">
|
||||
<reference anchor="Nigel79">
|
||||
<front>
|
||||
<title>Range encoding: An algorithm for removing redundancy from a digitised message</title>
|
||||
<author initials="G." surname="Nigel" fullname=""><organization/></author>
|
||||
|
@ -7654,6 +7667,20 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
|
|||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Vorbis-website" target="http://vorbis.com/">
|
||||
<front>
|
||||
<title>Vorbis website</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Matroska-website" target="http://matroska.org/">
|
||||
<front>
|
||||
<title>Matroska website</title>
|
||||
<author></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Vectors-website" target="http://opus-codec.org/testvectors/">
|
||||
<front>
|
||||
<title>Opus Testvectors (webside)</title>
|
||||
|
@ -7668,6 +7695,13 @@ Robust and Efficient Quantization of Speech LSP Parameters Using Structured Vect
|
|||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="range-coding" target="http://en.wikipedia.org/wiki/Range_coding">
|
||||
<front>
|
||||
<title>Range Coding</title>
|
||||
<author><organization>Wikipedia</organization></author>
|
||||
</front>
|
||||
</reference>
|
||||
|
||||
<reference anchor="Hadamard" target="http://en.wikipedia.org/wiki/Hadamard_transform">
|
||||
<front>
|
||||
<title>Hadamard Transform</title>
|
||||
|
|
Loading…
Add table
Add a link
Reference in a new issue