diff --git a/.opus.metadata b/.opus.metadata
new file mode 100644
index 0000000..e32ca83
--- /dev/null
+++ b/.opus.metadata
@@ -0,0 +1 @@
+37dd3d69b10612cd513ccf26675ef6d61eda24b4 SOURCES/opus-1.0.2.tar.gz
diff --git a/README.md b/README.md
deleted file mode 100644
index 0e7897f..0000000
--- a/README.md
+++ /dev/null
@@ -1,5 +0,0 @@
-The master branch has no content
- 
-Look at the c7 branch if you are working with CentOS-7, or the c4/c5/c6 branch for CentOS-4, 5 or 6
- 
-If you find this file in a distro specific branch, it means that no content has been checked in yet
diff --git a/SOURCES/0001-Fix-several-memory-errors-in-the-SILK-resampler.patch b/SOURCES/0001-Fix-several-memory-errors-in-the-SILK-resampler.patch
new file mode 100644
index 0000000..cdb005d
--- /dev/null
+++ b/SOURCES/0001-Fix-several-memory-errors-in-the-SILK-resampler.patch
@@ -0,0 +1,120 @@
+From dc58579c2c7e060084554018e9a2e8c25097a255 Mon Sep 17 00:00:00 2001
+From: "Timothy B. Terriberry" <tterribe@xiph.org>
+Date: Wed, 8 May 2013 10:25:52 -0700
+Subject: [PATCH] Fix several memory errors in the SILK resampler.
+
+1) The memcpy's were using sizeof(opus_int32), but the type of the
+    local buffer was opus_int16.
+2) Because the size was wrong, this potentially allowed the source
+    and destination regions of the memcpy overlap.
+   I _believe_ that nSamplesIn is at least fs_in_khZ, which is at
+    least 8.
+   Since RESAMPLER_ORDER_FIR_12 is only 8, I don't think that's a
+    problem once you fix the type size.
+3) The size of the buffer used RESAMPLER_MAX_BATCH_SIZE_IN, but the
+    data stored in it was actually _twice_ the input batch size
+    (nSamplesIn<<1).
+
+Because this never blew up in testing, I suspect that in practice
+ the batch sizes are reasonable enough that none of these things
+ was ever a problem, but proving that seems non-obvious.
+
+This patch just converts the whole thing to use CELT's vararrays.
+This fixes the buffer size problems (since we allocate a buffer
+ with the actual size we use) and gets these large buffers off the
+ stack on devices using the pseudo-stack.
+It also fixes the memcpy problems by changing the sizeof to
+ opus_int16.
+It turns out sFIR, which saved state between calls, was being used
+ elsewhere as opus_int32, so this converts it to a union to make
+ this sharing explicit.
+---
+ silk/resampler_private_IIR_FIR.c  | 14 +++++++++-----
+ silk/resampler_private_down_FIR.c |  4 ++--
+ silk/resampler_structs.h          |  5 ++++-
+ 3 files changed, 15 insertions(+), 8 deletions(-)
+
+diff --git a/silk/resampler_private_IIR_FIR.c b/silk/resampler_private_IIR_FIR.c
+index d9e42ca..2b9602d 100644
+--- a/silk/resampler_private_IIR_FIR.c
++++ b/silk/resampler_private_IIR_FIR.c
+@@ -31,6 +31,7 @@ POSSIBILITY OF SUCH DAMAGE.
+ 
+ #include "SigProc_FIX.h"
+ #include "resampler_private.h"
++#include "stack_alloc.h"
+ 
+ static inline opus_int16 *silk_resampler_private_IIR_FIR_INTERPOL(
+     opus_int16  *out,
+@@ -71,10 +72,13 @@ void silk_resampler_private_IIR_FIR(
+     silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
+     opus_int32 nSamplesIn;
+     opus_int32 max_index_Q16, index_increment_Q16;
+-    opus_int16 buf[ RESAMPLER_MAX_BATCH_SIZE_IN + RESAMPLER_ORDER_FIR_12 ];
++    VARDECL( opus_int16, buf );
++    SAVE_STACK;
++
++    ALLOC( buf, 2 * S->batchSize + RESAMPLER_ORDER_FIR_12, opus_int16 );
+ 
+     /* Copy buffered samples to start of buffer */
+-    silk_memcpy( buf, S->sFIR, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
++    silk_memcpy( buf, S->sFIR.i16, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
+ 
+     /* Iterate over blocks of frameSizeIn input samples */
+     index_increment_Q16 = S->invRatio_Q16;
+@@ -91,13 +95,13 @@ void silk_resampler_private_IIR_FIR(
+ 
+         if( inLen > 0 ) {
+             /* More iterations to do; copy last part of filtered signal to beginning of buffer */
+-            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
++            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
+         } else {
+             break;
+         }
+     }
+ 
+     /* Copy last part of filtered signal to the state for the next call */
+-    silk_memcpy( S->sFIR, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int32 ) );
++    silk_memcpy( S->sFIR.i16, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
++    RESTORE_STACK;
+ }
+-
+diff --git a/silk/resampler_private_down_FIR.c b/silk/resampler_private_down_FIR.c
+index 5d24564..8bedb0d 100644
+--- a/silk/resampler_private_down_FIR.c
++++ b/silk/resampler_private_down_FIR.c
+@@ -155,7 +155,7 @@ void silk_resampler_private_down_FIR(
+     const opus_int16 *FIR_Coefs;
+ 
+     /* Copy buffered samples to start of buffer */
+-    silk_memcpy( buf, S->sFIR, S->FIR_Order * sizeof( opus_int32 ) );
++    silk_memcpy( buf, S->sFIR.i32, S->FIR_Order * sizeof( opus_int32 ) );
+ 
+     FIR_Coefs = &S->Coefs[ 2 ];
+ 
+@@ -185,5 +185,5 @@ void silk_resampler_private_down_FIR(
+     }
+ 
+     /* Copy last part of filtered signal to the state for the next call */
+-    silk_memcpy( S->sFIR, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
++    silk_memcpy( S->sFIR.i32, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
+ }
+diff --git a/silk/resampler_structs.h b/silk/resampler_structs.h
+index 4c28bd0..d1a0b95 100644
+--- a/silk/resampler_structs.h
++++ b/silk/resampler_structs.h
+@@ -37,7 +37,10 @@ extern "C" {
+ 
+ typedef struct _silk_resampler_state_struct{
+     opus_int32       sIIR[ SILK_RESAMPLER_MAX_IIR_ORDER ]; /* this must be the first element of this struct */
+-    opus_int32       sFIR[ SILK_RESAMPLER_MAX_FIR_ORDER ];
++    union{
++        opus_int32   i32[ SILK_RESAMPLER_MAX_FIR_ORDER ];
++        opus_int16   i16[ SILK_RESAMPLER_MAX_FIR_ORDER ];
++    }                sFIR;
+     opus_int16       delayBuf[ 48 ];
+     opus_int         resampler_function;
+     opus_int         batchSize;
+-- 
+1.8.4.2
+
diff --git a/SOURCES/0001-Fixes-an-assertion-failure-in-SILK.patch b/SOURCES/0001-Fixes-an-assertion-failure-in-SILK.patch
new file mode 100644
index 0000000..cb8f5e5
--- /dev/null
+++ b/SOURCES/0001-Fixes-an-assertion-failure-in-SILK.patch
@@ -0,0 +1,74 @@
+From ac76b1503f759201f03dc6acb7bf00bd39f560d3 Mon Sep 17 00:00:00 2001
+From: Jean-Marc Valin <jmvalin@jmvalin.ca>
+Date: Thu, 9 May 2013 16:17:13 -0400
+Subject: [PATCH] Fixes an assertion failure in SILK
+
+We stop the schur recursion before any reflection coefficient
+goes outside of ]-1,1[ and we force reporting a residual energy
+of at least 1.
+Assertion was:
+Fatal (internal) error in ../silk/fixed/noise_shape_analysis_FIX.c, line 290: assertion failed: nrg >= 0
+triggered by:
+opus_demo voip 16000 1 12500 -bandwidth WB -complexity 10 pl04f087.stp-crash out.pcm
+---
+ silk/fixed/schur64_FIX.c | 11 ++++++++++-
+ silk/fixed/schur_FIX.c   | 10 +++++++++-
+ 2 files changed, 19 insertions(+), 2 deletions(-)
+
+diff --git a/silk/fixed/schur64_FIX.c b/silk/fixed/schur64_FIX.c
+index 5ff2756..c75f96a 100644
+--- a/silk/fixed/schur64_FIX.c
++++ b/silk/fixed/schur64_FIX.c
+@@ -56,6 +56,11 @@ opus_int32 silk_schur64(                            /* O    returns residual ene
+     }
+ 
+     for( k = 0; k < order; k++ ) {
++        /* Check that we won't be getting an unstable rc, otherwise stop here. */
++        if (silk_abs_int32(C[ k + 1 ][ 0 ]) >= C[ 0 ][ 1 ]) {
++           break;
++        }
++
+         /* Get reflection coefficient: divide two Q30 values and get result in Q31 */
+         rc_tmp_Q31 = silk_DIV32_varQ( -C[ k + 1 ][ 0 ], C[ 0 ][ 1 ], 31 );
+ 
+@@ -73,5 +78,9 @@ opus_int32 silk_schur64(                            /* O    returns residual ene
+         }
+     }
+ 
+-    return( C[ 0 ][ 1 ] );
++    for(; k < order; k++ ) {
++       rc_Q16[ k ] = 0;
++    }
++
++    return silk_max_32( 1, C[ 0 ][ 1 ] );
+ }
+diff --git a/silk/fixed/schur_FIX.c b/silk/fixed/schur_FIX.c
+index 43db501..788ad3f 100644
+--- a/silk/fixed/schur_FIX.c
++++ b/silk/fixed/schur_FIX.c
+@@ -68,6 +68,10 @@ opus_int32 silk_schur(                              /* O    Returns residual ene
+     }
+ 
+     for( k = 0; k < order; k++ ) {
++        /* Check that we won't be getting an unstable rc, otherwise stop here. */
++        if (silk_abs_int32(C[ k + 1 ][ 0 ]) >= C[ 0 ][ 1 ]) {
++           break;
++        }
+ 
+         /* Get reflection coefficient */
+         rc_tmp_Q15 = -silk_DIV32_16( C[ k + 1 ][ 0 ], silk_max_32( silk_RSHIFT( C[ 0 ][ 1 ], 15 ), 1 ) );
+@@ -87,6 +91,10 @@ opus_int32 silk_schur(                              /* O    Returns residual ene
+         }
+     }
+ 
++    for(; k < order; k++ ) {
++       rc_Q15[ k ] = 0;
++    }
++
+     /* return residual energy */
+-    return C[ 0 ][ 1 ];
++    return silk_max_32( 1, C[ 0 ][ 1 ] );
+ }
+-- 
+1.8.4.2
+
diff --git a/SOURCES/rfc6716.txt b/SOURCES/rfc6716.txt
new file mode 100644
index 0000000..21b690f
--- /dev/null
+++ b/SOURCES/rfc6716.txt
@@ -0,0 +1,18259 @@
+
+
+
+
+
+
+Internet Engineering Task Force (IETF)                         JM. Valin
+Request for Comments: 6716                           Mozilla Corporation
+Category: Standards Track                                         K. Vos
+ISSN: 2070-1721                                  Skype Technologies S.A.
+                                                           T. Terriberry
+                                                     Mozilla Corporation
+                                                          September 2012
+
+
+                   Definition of the Opus Audio Codec
+
+Abstract
+
+   This document defines the Opus interactive speech and audio codec.
+   Opus is designed to handle a wide range of interactive audio
+   applications, including Voice over IP, videoconferencing, in-game
+   chat, and even live, distributed music performances.  It scales from
+   low bitrate narrowband speech at 6 kbit/s to very high quality stereo
+   music at 510 kbit/s.  Opus uses both Linear Prediction (LP) and the
+   Modified Discrete Cosine Transform (MDCT) to achieve good compression
+   of both speech and music.
+
+Status of This Memo
+
+   This is an Internet Standards Track document.
+
+   This document is a product of the Internet Engineering Task Force
+   (IETF).  It represents the consensus of the IETF community.  It has
+   received public review and has been approved for publication by the
+   Internet Engineering Steering Group (IESG).  Further information on
+   Internet Standards is available in Section 2 of RFC 5741.
+
+   Information about the current status of this document, any errata,
+   and how to provide feedback on it may be obtained at
+   http://www.rfc-editor.org/info/rfc6716.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                    [Page 1]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+Copyright Notice
+
+   Copyright (c) 2012 IETF Trust and the persons identified as the
+   document authors.  All rights reserved.
+
+   This document is subject to BCP 78 and the IETF Trust's Legal
+   Provisions Relating to IETF Documents
+   (http://trustee.ietf.org/license-info) in effect on the date of
+   publication of this document.  Please review these documents
+   carefully, as they describe your rights and restrictions with respect
+   to this document.  Code Components extracted from this document must
+   include Simplified BSD License text as described in Section 4.e of
+   the Trust Legal Provisions and are provided without warranty as
+   described in the Simplified BSD License.
+
+   The licenses granted by the IETF Trust to this RFC under Section 3.c
+   of the Trust Legal Provisions shall also include the right to extract
+   text from Sections 1 through 8 and Appendix A and Appendix B of this
+   RFC and create derivative works from these extracts, and to copy,
+   publish, display and distribute such derivative works in any medium
+   and for any purpose, provided that no such derivative work shall be
+   presented, displayed or published in a manner that states or implies
+   that it is part of this RFC or any other IETF Document.
+
+Table of Contents
+
+   1. Introduction ....................................................5
+      1.1. Notation and Conventions ...................................6
+   2. Opus Codec Overview .............................................8
+      2.1. Control Parameters ........................................10
+           2.1.1. Bitrate ............................................10
+           2.1.2. Number of Channels (Mono/Stereo) ...................11
+           2.1.3. Audio Bandwidth ....................................11
+           2.1.4. Frame Duration .....................................11
+           2.1.5. Complexity .........................................11
+           2.1.6. Packet Loss Resilience .............................12
+           2.1.7. Forward Error Correction (FEC) .....................12
+           2.1.8. Constant/Variable Bitrate ..........................12
+           2.1.9. Discontinuous Transmission (DTX) ...................13
+   3. Internal Framing ...............................................13
+      3.1. The TOC Byte ..............................................13
+      3.2. Frame Packing .............................................16
+           3.2.1. Frame Length Coding ................................16
+           3.2.2. Code 0: One Frame in the Packet ....................16
+           3.2.3. Code 1: Two Frames in the Packet, Each with
+                  Equal Compressed Size ..............................17
+           3.2.4. Code 2: Two Frames in the Packet, with
+                  Different Compressed Sizes .........................17
+
+
+
+Valin, et al.                Standards Track                    [Page 2]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+           3.2.5. Code 3: A Signaled Number of Frames in the Packet ..18
+      3.3. Examples ..................................................21
+      3.4. Receiving Malformed Packets ...............................22
+   4. Opus Decoder ...................................................23
+      4.1. Range Decoder .............................................23
+           4.1.1. Range Decoder Initialization .......................25
+           4.1.2. Decoding Symbols ...................................25
+           4.1.3. Alternate Decoding Methods .........................27
+           4.1.4. Decoding Raw Bits ..................................29
+           4.1.5. Decoding Uniformly Distributed Integers ............29
+           4.1.6. Current Bit Usage ..................................30
+      4.2. SILK Decoder ..............................................32
+           4.2.1. SILK Decoder Modules ...............................32
+           4.2.2. LP Layer Organization ..............................33
+           4.2.3. Header Bits ........................................35
+           4.2.4. Per-Frame LBRR Flags ...............................36
+           4.2.5. LBRR Frames ........................................36
+           4.2.6. Regular SILK Frames ................................37
+           4.2.7. SILK Frame Contents ................................37
+                  4.2.7.1. Stereo Prediction Weights .................40
+                  4.2.7.2. Mid-Only Flag .............................42
+                  4.2.7.3. Frame Type ................................43
+                  4.2.7.4. Subframe Gains ............................44
+                  4.2.7.5. Normalized Line Spectral Frequency
+                           (LSF) and Linear Predictive Coding (LPC)
+                           Coeffieients ..............................46
+                  4.2.7.6. Long-Term Prediction (LTP) Parameters .....74
+                  4.2.7.7. Linear Congruential Generator (LCG) Seed ..86
+                  4.2.7.8. Excitation ................................86
+                  4.2.7.9. SILK Frame Reconstruction .................98
+           4.2.8. Stereo Unmixing ...................................102
+           4.2.9. Resampling ........................................103
+      4.3. CELT Decoder .............................................104
+           4.3.1. Transient Decoding ................................108
+           4.3.2. Energy Envelope Decoding ..........................108
+           4.3.3. Bit Allocation ....................................110
+           4.3.4. Shape Decoding ....................................116
+           4.3.5. Anti-collapse Processing ..........................120
+           4.3.6. Denormalization ...................................121
+           4.3.7. Inverse MDCT ......................................121
+      4.4. Packet Loss Concealment (PLC) ............................122
+           4.4.1. Clock Drift Compensation ..........................122
+      4.5. Configuration Switching ..................................123
+           4.5.1. Transition Side Information (Redundancy) ..........124
+           4.5.2. State Reset .......................................127
+           4.5.3. Summary of Transitions ............................128
+   5. Opus Encoder ..................................................131
+      5.1. Range Encoder ............................................132
+
+
+
+Valin, et al.                Standards Track                    [Page 3]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+           5.1.1. Encoding Symbols ..................................133
+           5.1.2. Alternate Encoding Methods ........................134
+           5.1.3. Encoding Raw Bits .................................135
+           5.1.4. Encoding Uniformly Distributed Integers ...........135
+           5.1.5. Finalizing the Stream .............................135
+           5.1.6. Current Bit Usage .................................136
+      5.2. SILK Encoder .............................................136
+           5.2.1. Sample Rate Conversion ............................137
+           5.2.2. Stereo Mixing .....................................137
+           5.2.3. SILK Core Encoder .................................138
+      5.3. CELT Encoder .............................................150
+           5.3.1. Pitch Pre-filter ..................................150
+           5.3.2. Bands and Normalization ...........................151
+           5.3.3. Energy Envelope Quantization ......................151
+           5.3.4. Bit Allocation ....................................151
+           5.3.5. Stereo Decisions ..................................152
+           5.3.6. Time-Frequency Decision ...........................153
+           5.3.7. Spreading Values Decision .........................153
+           5.3.8. Spherical Vector Quantization .....................154
+   6. Conformance ...................................................155
+      6.1. Testing ..................................................155
+      6.2. Opus Custom ..............................................156
+   7. Security Considerations .......................................157
+   8. Acknowledgements ..............................................158
+   9. References ....................................................159
+      9.1. Normative References .....................................159
+      9.2. Informative References ...................................159
+   Appendix A. Reference Implementation .............................163
+      A.1. Extracting the Source ....................................164
+      A.2. Up-to-Date Implementation ................................164
+      A.3. Base64-Encoded Source Code ...............................164
+      A.4. Test Vectors .............................................321
+   Appendix B. Self-Delimiting Framing ..............................321
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                    [Page 4]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+1.  Introduction
+
+   The Opus codec is a real-time interactive audio codec designed to
+   meet the requirements described in [REQUIREMENTS].  It is composed of
+   a layer based on Linear Prediction (LP) [LPC] and a layer based on
+   the Modified Discrete Cosine Transform (MDCT) [MDCT].  The main idea
+   behind using two layers is as follows: in speech, linear prediction
+   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 either one alone and can achieve
+   better quality by combining them than by using either one
+   individually.
+
+   The primary normative part of this specification is provided by the
+   source code in Appendix A.  Only the decoder portion of this software
+   is normative, though a significant amount of code is shared by both
+   the encoder and decoder.  Section 6 provides a decoder conformance
+   test.  The decoder contains a great deal of integer and fixed-point
+   arithmetic that needs to be performed exactly, including all rounding
+   considerations, so any useful specification requires domain-specific
+   symbolic language to adequately define these operations.
+   Additionally, any conflict between the symbolic representation and
+   the included reference implementation must be resolved.  For the
+   practical reasons of compatibility and testability, it would be
+   advantageous to give the reference implementation priority in any
+   disagreement.  The C language is also one of the most widely
+   understood, human-readable symbolic representations for machine
+   behavior.  For these reasons, this RFC uses the reference
+   implementation as the sole symbolic representation of the codec.
+
+   While the symbolic representation is unambiguous and complete, it is
+   not always the easiest way to understand the codec's operation.  For
+   this reason, this document also describes significant parts of the
+   codec in prose and takes the opportunity to explain the rationale
+   behind many of the more surprising elements of the design.  These
+   descriptions are intended to be accurate and informative, but the
+   limitations of common English sometimes result in ambiguity, so it is
+   expected that the reader will always read them alongside the symbolic
+   representation.  Numerous references to the implementation are
+   provided for this purpose.  The descriptions sometimes differ from
+   the reference in ordering or through mathematical simplification
+   wherever such deviation makes an explanation easier to understand.
+   For example, the right shift and left shift operations in the
+   reference implementation are often described using division and
+
+
+
+
+
+Valin, et al.                Standards Track                    [Page 5]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   multiplication in the text.  In general, the text is focused on the
+   "what" and "why" while the symbolic representation most clearly
+   provides the "how".
+
+1.1.  Notation and Conventions
+
+   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
+   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
+   document are to be interpreted as described in RFC 2119 [RFC2119].
+
+   Various operations in the codec require bit-exact fixed-point
+   behavior, even when writing a floating point implementation.  The
+   notation "Q<n>", where n is an integer, denotes the number of binary
+   digits to the right of the decimal point in a fixed-point number.
+   For example, a signed Q14 value in a 16-bit word can represent values
+   from -2.0 to 1.99993896484375, inclusive.  This notation is for
+   informational purposes only.  Arithmetic, when described, always
+   operates on the underlying integer.  For example, the text will
+   explicitly indicate any shifts required after a multiplication.
+
+   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 text also makes use of the following
+   functions.
+
+1.1.1.  min(x,y)
+
+   The smallest of two values x and y.
+
+1.1.2.  max(x,y)
+
+   The largest of two values x and y.
+
+1.1.3.  clamp(lo,x,hi)
+
+                     clamp(lo,x,hi) = max(lo,min(x,hi))
+
+   With this definition, if lo > hi, then lo is returned.
+
+1.1.4.  sign(x)
+
+   The sign of x, i.e.,
+
+                                    ( -1,  x < 0
+                          sign(x) = <  0,  x == 0
+                                    (  1,  x > 0
+
+
+
+
+
+Valin, et al.                Standards Track                    [Page 6]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+1.1.5.  abs(x)
+
+   The absolute value of x, i.e.,
+
+                             abs(x) = sign(x)*x
+
+1.1.6.  floor(f)
+
+   The largest integer z such that z <= f.
+
+1.1.7.  ceil(f)
+
+   The smallest integer z such that z >= f.
+
+1.1.8.  round(f)
+
+   The integer z nearest to f, with ties rounded towards negative
+   infinity, i.e.,
+
+                           round(f) = ceil(f - 0.5)
+
+1.1.9.  log2(f)
+
+   The base-two logarithm of f.
+
+1.1.10.  ilog(n)
+
+   The minimum number of bits required to store a positive integer n in
+   binary, or 0 for a non-positive integer n.
+
+                              ( 0,                 n <= 0
+                    ilog(n) = <
+                              ( floor(log2(n))+1,  n > 0
+
+   Examples:
+
+   o  ilog(-1) = 0
+
+   o  ilog(0) = 0
+
+   o  ilog(1) = 1
+
+   o  ilog(2) = 2
+
+   o  ilog(3) = 2
+
+   o  ilog(4) = 3
+
+
+
+
+Valin, et al.                Standards Track                    [Page 7]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  ilog(7) = 3
+
+2.  Opus Codec Overview
+
+   The Opus codec scales from 6 kbit/s narrowband mono speech to
+   510 kbit/s fullband stereo music, with algorithmic delays ranging
+   from 5 ms to 65.2 ms.  At any given time, either the LP layer, the
+   MDCT layer, or both, may be active.  It can seamlessly switch between
+   all of its various operating modes, giving it a great deal of
+   flexibility to adapt to varying content and network conditions
+   without renegotiating the current session.  The codec allows input
+   and output of various audio bandwidths, defined as follows:
+
+   +----------------------+-----------------+-------------------------+
+   | Abbreviation         | Audio Bandwidth | Sample Rate (Effective) |
+   +----------------------+-----------------+-------------------------+
+   | NB (narrowband)      |           4 kHz |                   8 kHz |
+   |                      |                 |                         |
+   | MB (medium-band)     |           6 kHz |                  12 kHz |
+   |                      |                 |                         |
+   | WB (wideband)        |           8 kHz |                  16 kHz |
+   |                      |                 |                         |
+   | SWB (super-wideband) |          12 kHz |                  24 kHz |
+   |                      |                 |                         |
+   | FB (fullband)        |      20 kHz (*) |                  48 kHz |
+   +----------------------+-----------------+-------------------------+
+
+                                  Table 1
+
+   (*) Although the sampling theorem allows a bandwidth as large as half
+   the sampling rate, Opus never codes audio above 20 kHz, as that is
+   the generally accepted upper limit of human hearing.
+
+   Opus defines super-wideband (SWB) with an effective sample rate of
+   24 kHz, unlike some other audio coding standards that use 32 kHz.
+   This was chosen for a number of reasons.  The band layout in the MDCT
+   layer naturally allows skipping coefficients for frequencies over
+   12 kHz, but does not allow cleanly dropping just those frequencies
+   over 16 kHz.  A sample rate of 24 kHz also makes resampling in the
+   MDCT layer easier, as 24 evenly divides 48, and when 24 kHz is
+   sufficient, it can save computation in other processing, such as
+   Acoustic Echo Cancellation (AEC).  Experimental changes to the band
+   layout to allow a 16 kHz cutoff (32 kHz effective sample rate) showed
+   potential quality degradations at other sample rates, and, at typical
+   bitrates, the number of bits saved by using such a cutoff instead of
+   coding in fullband (FB) mode is very small.  Therefore, if an
+   application wishes to process a signal sampled at 32 kHz, it should
+   just use FB.
+
+
+
+Valin, et al.                Standards Track                    [Page 8]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The LP layer is based on the SILK codec [SILK].  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 [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 compatible with, the stand-alone SILK codec previously
+   deployed by Skype.  This document does not serve to define that
+   format, but those interested in the original SILK codec should see
+   [SILK] instead.
+
+   The MDCT layer is based on the Constrained-Energy Lapped Transform
+   (CELT) codec [CELT].  It supports NB, WB, SWB, or FB audio and frame
+   sizes from 2.5 ms to 20 ms, and requires an additional 2.5 ms look-
+   ahead due to the overlapping MDCT windows.  The CELT codec is
+   inherently designed for CBR coding, but unlike many CBR codecs, it is
+   not limited to a set of predetermined rates.  It internally allocates
+   bits to exactly fill any given target budget, and an encoder can
+   produce a VBR stream by varying the target on a per-frame basis.  The
+   MDCT layer is not used for speech when the audio bandwidth is WB or
+   less, as it is not useful there.  On the other hand, non-speech
+   signals are not always adequately coded using linear prediction.
+   Therefore, the MDCT layer should be used for music signals.
+
+   A "Hybrid" mode allows the use of both layers simultaneously with a
+   frame size of 10 or 20 ms and an SWB or FB audio bandwidth.  The LP
+   layer codes the low frequencies by resampling the signal down to WB.
+   The MDCT layer follows, coding the high frequency portion of the
+   signal.  The cutoff between the two lies at 8 kHz, the maximum WB
+   audio bandwidth.  In the MDCT layer, all bands below 8 kHz are
+   discarded, so there is no coding redundancy between the two layers.
+
+   The sample rate (in contrast to the actual audio bandwidth) can be
+   chosen independently on the encoder and decoder side, e.g., a
+   fullband signal can be decoded as wideband, or vice versa.  This
+   approach ensures a sender and receiver can always interoperate,
+   regardless of the capabilities of their actual audio hardware.
+   Internally, the LP layer always operates at a sample rate of twice
+   the audio bandwidth, up to a maximum of 16 kHz, which it continues to
+   use for SWB and FB.  The decoder simply resamples its output to
+   support different sample rates.  The MDCT layer always operates
+   internally at a sample rate of 48 kHz.  Since all the supported
+   sample rates evenly divide this rate, and since the decoder may
+   easily zero out the high frequency portion of the spectrum in the
+   frequency domain, it can simply decimate the MDCT layer output to
+   achieve the other supported sample rates very cheaply.
+
+
+
+Valin, et al.                Standards Track                    [Page 9]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   After conversion to the common, desired output sample rate, the
+   decoder simply adds the output from the two layers together.  To
+   compensate for the different look-ahead required by each layer, the
+   CELT encoder input is delayed by an additional 2.7 ms.  This ensures
+   that low frequencies and high frequencies arrive at the same time.
+   This extra delay may be reduced by an encoder by using less look-
+   ahead for noise shaping or using a simpler resampler in the LP layer,
+   but this will reduce quality.  However, the base 2.5 ms look-ahead in
+   the CELT layer cannot be reduced in the encoder because it is needed
+   for the MDCT overlap, whose size is fixed by the decoder.
+
+   Both layers use the same entropy coder, avoiding any waste from
+   "padding bits" between them.  The hybrid approach makes it easy to
+   support both CBR and VBR coding.  Although the LP layer is VBR, the
+   bit allocation of the MDCT layer can produce a final stream that is
+   CBR by using all the bits left unused by the LP layer.
+
+2.1.  Control Parameters
+
+   The Opus codec includes a number of control parameters that can be
+   changed dynamically during regular operation of the codec, without
+   interrupting the audio stream from the encoder to the decoder.  These
+   parameters only affect the encoder since any impact they have on the
+   bitstream is signaled in-band such that a decoder can decode any Opus
+   stream without any out-of-band signaling.  Any Opus implementation
+   can add or modify these control parameters without affecting
+   interoperability.  The most important encoder control parameters in
+   the reference encoder are listed below.
+
+2.1.1.  Bitrate
+
+   Opus supports all bitrates from 6 kbit/s to 510 kbit/s.  All other
+   parameters being equal, higher bitrate results in higher quality.
+   For a frame size of 20 ms, these are the bitrate "sweet spots" for
+   Opus in various configurations:
+
+   o  8-12 kbit/s for NB speech,
+
+   o  16-20 kbit/s for WB speech,
+
+   o  28-40 kbit/s for FB speech,
+
+   o  48-64 kbit/s for FB mono music, and
+
+   o  64-128 kbit/s for FB stereo music.
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 10]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+2.1.2.  Number of Channels (Mono/Stereo)
+
+   Opus can transmit either mono or stereo frames within a single
+   stream.  When decoding a mono frame in a stereo decoder, the left and
+   right channels are identical, and when decoding a stereo frame in a
+   mono decoder, the mono output is the average of the left and right
+   channels.  In some cases, it is desirable to encode a stereo input
+   stream in mono (e.g., because the bitrate is too low to encode stereo
+   with sufficient quality).  The number of channels encoded can be
+   selected in real-time, but by default the reference encoder attempts
+   to make the best decision possible given the current bitrate.
+
+2.1.3.  Audio Bandwidth
+
+   The audio bandwidths supported by Opus are listed in Table 1.  Just
+   like for the number of channels, any decoder can decode audio that is
+   encoded at any bandwidth.  For example, any Opus decoder operating at
+   8 kHz can decode an FB Opus frame, and any Opus decoder operating at
+   48 kHz can decode an NB frame.  Similarly, the reference encoder can
+   take a 48 kHz input signal and encode it as NB.  The higher the audio
+   bandwidth, the higher the required bitrate to achieve acceptable
+   quality.  The audio bandwidth can be explicitly specified in real-
+   time, but, by default, the reference encoder attempts to make the
+   best bandwidth decision possible given the current bitrate.
+
+2.1.4.  Frame Duration
+
+   Opus can encode frames of 2.5, 5, 10, 20, 40, or 60 ms.  It can also
+   combine multiple frames into packets of up to 120 ms.  For real-time
+   applications, sending fewer packets per second reduces the bitrate,
+   since it reduces the overhead from IP, UDP, and RTP headers.
+   However, it increases latency and sensitivity to packet losses, as
+   losing one packet constitutes a loss of a bigger chunk of audio.
+   Increasing the frame duration also slightly improves coding
+   efficiency, but the gain becomes small for frame sizes above 20 ms.
+   For this reason, 20 ms frames are a good choice for most
+   applications.
+
+2.1.5.  Complexity
+
+   There are various aspects of the Opus encoding process where trade-
+   offs can be made between CPU complexity and quality/bitrate.  In the
+   reference encoder, the complexity is selected using an integer from 0
+   to 10, where 0 is the lowest complexity and 10 is the highest.
+   Examples of computations for which such trade-offs may occur are:
+
+   o  The order of the pitch analysis whitening filter [WHITENING],
+
+
+
+
+Valin, et al.                Standards Track                   [Page 11]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  The order of the short-term noise shaping filter,
+
+   o  The number of states in delayed decision quantization of the
+      residual signal, and
+
+   o  The use of certain bitstream features such as variable time-
+      frequency resolution and the pitch post-filter.
+
+2.1.6.  Packet Loss Resilience
+
+   Audio codecs often exploit inter-frame correlations to reduce the
+   bitrate at a cost in error propagation: after losing one packet,
+   several packets need to be received before the decoder is able to
+   accurately reconstruct the speech signal.  The extent to which Opus
+   exploits inter-frame dependencies can be adjusted on the fly to
+   choose a trade-off between bitrate and amount of error propagation.
+
+2.1.7.  Forward Error Correction (FEC)
+
+   Another mechanism providing robustness against packet loss is the in-
+   band Forward Error Correction (FEC).  Packets that are determined to
+   contain perceptually important speech information, such as onsets or
+   transients, are encoded again at a lower bitrate and this re-encoded
+   information is added to a subsequent packet.
+
+2.1.8.  Constant/Variable Bitrate
+
+   Opus is more efficient when operating with variable bitrate (VBR),
+   which is the default.  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 (MPEG 1, Layer 3) and AAC (Advanced Audio
+   Coding) call CBR (i.e., not true CBR due to the bit reservoir).  In
+   some (rare) applications, constant bitrate (CBR) is required.  There
+   are two main reasons to operate in CBR mode:
+
+   o  When the transport only supports a fixed size for each compressed
+      frame, or
+
+   o  When encryption is used for an audio stream that is either highly
+      constrained (e.g., yes/no, recorded prompts) or highly sensitive
+      [SRTP-VBR].
+
+   Bitrate may still be allowed to vary, even with sensitive data, as
+   long as the variation is not driven by the input signal (for example,
+   to match changing network conditions).  To achieve this, an
+   application should still run Opus in CBR mode, but change the target
+   rate before each packet.
+
+
+
+Valin, et al.                Standards Track                   [Page 12]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+2.1.9.  Discontinuous Transmission (DTX)
+
+   Discontinuous Transmission (DTX) reduces the bitrate during silence
+   or background noise.  When DTX is enabled, only one frame is encoded
+   every 400 milliseconds.
+
+3.  Internal Framing
+
+   The Opus encoder produces "packets", which are each a contiguous set
+   of bytes meant to be transmitted as a single unit.  The packets
+   described here do not include such things as IP, UDP, or RTP headers,
+   which are normally found in a transport-layer packet.  A single
+   packet may contain multiple audio frames, so long as they share a
+   common set of parameters, including the operating mode, audio
+   bandwidth, frame size, and channel count (mono vs. stereo).  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 lower layer (such as UDP or RTP [RFC3550] or Ogg [RFC3533] or
+   Matroska [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.  An alternative, self-
+   delimiting variant of the framing is described in Appendix B.
+   Support for that variant is OPTIONAL.
+
+   All bit diagrams in this document number the bits so that bit 0 is
+   the most significant bit of the first byte, and bit 7 is the least
+   significant.  Bit 8 is thus the most significant bit of the second
+   byte, etc.  Well-formed Opus packets obey certain requirements,
+   marked [R1] through [R7] below.  These are summarized in Section 3.4
+   along with appropriate means of handling malformed packets.
+
+3.1.  The TOC Byte
+
+   A well-formed Opus packet MUST contain at least one byte [R1].  This
+   byte forms a table-of-contents (TOC) header that signals which of the
+   various modes and configurations a given packet uses.  It is composed
+   of a configuration number, "config", a stereo flag, "s", and a frame
+   count code, "c", arranged as illustrated in Figure 1.  A description
+   of each of these fields follows.
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 13]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                              0
+                              0 1 2 3 4 5 6 7
+                             +-+-+-+-+-+-+-+-+
+                             | config  |s| c |
+                             +-+-+-+-+-+-+-+-+
+
+                          Figure 1: The TOC Byte
+
+   The top five bits of the TOC byte, labeled "config", encode one of 32
+   possible configurations of operating mode, audio bandwidth, and frame
+   size.  As described, the LP (SILK) layer and MDCT (CELT) layer can be
+   combined in three possible operating modes:
+
+   1.  A SILK-only mode for use in low bitrate connections with an audio
+       bandwidth of WB or less,
+
+   2.  A Hybrid (SILK+CELT) mode for SWB or FB speech at medium
+       bitrates, and
+
+   3.  A CELT-only mode for very low delay speech transmission as well
+       as music transmission (NB to FB).
+
+   The 32 possible configurations each identify which one of these
+   operating modes the packet uses, as well as the audio bandwidth and
+   the frame size.  Table 2 lists the parameters for each configuration.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 14]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +-----------------------+-----------+-----------+-------------------+
+   | Configuration         | Mode      | Bandwidth | Frame Sizes       |
+   | Number(s)             |           |           |                   |
+   +-----------------------+-----------+-----------+-------------------+
+   | 0...3                 | SILK-only | NB        | 10, 20, 40, 60 ms |
+   |                       |           |           |                   |
+   | 4...7                 | SILK-only | MB        | 10, 20, 40, 60 ms |
+   |                       |           |           |                   |
+   | 8...11                | SILK-only | WB        | 10, 20, 40, 60 ms |
+   |                       |           |           |                   |
+   | 12...13               | Hybrid    | SWB       | 10, 20 ms         |
+   |                       |           |           |                   |
+   | 14...15               | Hybrid    | FB        | 10, 20 ms         |
+   |                       |           |           |                   |
+   | 16...19               | CELT-only | NB        | 2.5, 5, 10, 20 ms |
+   |                       |           |           |                   |
+   | 20...23               | CELT-only | WB        | 2.5, 5, 10, 20 ms |
+   |                       |           |           |                   |
+   | 24...27               | CELT-only | SWB       | 2.5, 5, 10, 20 ms |
+   |                       |           |           |                   |
+   | 28...31               | CELT-only | FB        | 2.5, 5, 10, 20 ms |
+   +-----------------------+-----------+-----------+-------------------+
+
+                Table 2: TOC Byte Configuration Parameters
+
+   The configuration numbers in each range (e.g., 0...3 for NB SILK-
+   only) correspond to the various choices of frame size, in the same
+   order.  For example, configuration 0 has a 10 ms frame size and
+   configuration 3 has a 60 ms frame size.
+
+   One additional bit, labeled "s", signals mono vs. stereo, with 0
+   indicating mono and 1 indicating stereo.
+
+   The remaining two bits of the TOC byte, labeled "c", code the number
+   of frames per packet (codes 0 to 3) as follows:
+
+   o  0: 1 frame in the packet
+
+   o  1: 2 frames in the packet, each with equal compressed size
+
+   o  2: 2 frames in the packet, with different compressed sizes
+
+   o  3: an arbitrary number of frames in the packet
+
+   This document refers to a packet as a code 0 packet, code 1 packet,
+   etc., based on the value of "c".
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 15]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+3.2.  Frame Packing
+
+   This section describes how frames are packed according to each
+   possible value of "c" in the TOC byte.
+
+3.2.1.  Frame Length Coding
+
+   When a packet contains multiple VBR frames (i.e., code 2 or 3), the
+   compressed length of one or more of these frames is indicated with a
+   one- or two-byte sequence, with the meaning of the first byte as
+   follows:
+
+   o  0: No frame (Discontinuous Transmission (DTX) or lost packet)
+
+   o  1...251: Length of the frame in bytes
+
+   o  252...255: A second byte is needed.  The total length is
+      (second_byte*4)+first_byte
+
+   The special length 0 indicates that no frame is available, either
+   because it was dropped during transmission by some intermediary or
+   because the encoder chose not to transmit it.  Any Opus frame in any
+   mode MAY have a length of 0.
+
+   The maximum representable length is 255*4+255=1275 bytes.  For 20 ms
+   frames, this represents a bitrate of 510 kbit/s, which is
+   approximately the highest useful rate for lossily compressed fullband
+   stereo music.  Beyond this point, lossless codecs are more
+   appropriate.  It is also roughly the maximum useful rate of the MDCT
+   layer as, shortly thereafter, quality no longer improves with
+   additional bits due to limitations on the codebook sizes.
+
+   No length is transmitted for the last frame in a VBR packet, or for
+   any of the frames in a CBR packet, as it can be inferred from the
+   total size of the packet and the size of all other data in the
+   packet.  However, the length of any individual frame MUST NOT exceed
+   1275 bytes [R2] to allow for repacketization by gateways, conference
+   bridges, or other software.
+
+3.2.2.  Code 0: One Frame in the Packet
+
+   For code 0 packets, the TOC byte is immediately followed by N-1 bytes
+   of compressed data for a single frame (where N is the size of the
+   packet), as illustrated in Figure 2.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 16]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|0|0|                                               |
+     +-+-+-+-+-+-+-+-+                                               |
+     |                    Compressed frame 1 (N-1 bytes)...          :
+     :                                                               |
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                         Figure 2: A Code 0 Packet
+
+3.2.3.  Code 1: Two Frames in the Packet, Each with Equal Compressed
+        Size
+
+   For code 1 packets, the TOC byte is immediately followed by the
+   (N-1)/2 bytes of compressed data for the first frame, followed by
+   (N-1)/2 bytes of compressed data for the second frame, as illustrated
+   in Figure 3.  The number of payload bytes available for compressed
+   data, N-1, MUST be even for all code 1 packets [R3].
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|0|1|                                               |
+     +-+-+-+-+-+-+-+-+                                               :
+     |             Compressed frame 1 ((N-1)/2 bytes)...             |
+     :                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                               |                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               :
+     |             Compressed frame 2 ((N-1)/2 bytes)...             |
+     :                                               +-+-+-+-+-+-+-+-+
+     |                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                         Figure 3: A Code 1 Packet
+
+3.2.4.  Code 2: Two Frames in the Packet, with Different Compressed
+        Sizes
+
+   For code 2 packets, the TOC byte is followed by a one- or two-byte
+   sequence indicating the length of the first frame (marked N1 in
+   Figure 4), followed by N1 bytes of compressed data for the first
+   frame.  The remaining N-N1-2 or N-N1-3 bytes are the compressed data
+   for the second frame.  This is illustrated in Figure 4.  A code 2
+   packet MUST contain enough bytes to represent a valid length.  For
+   example, a 1-byte code 2 packet is always invalid, and a 2-byte code
+   2 packet whose second byte is in the range 252...255 is also invalid.
+
+
+
+Valin, et al.                Standards Track                   [Page 17]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The length of the first frame, N1, MUST also be no larger than the
+   size of the payload remaining after decoding that length for all code
+   2 packets [R4].  This makes, for example, a 2-byte code 2 packet with
+   a second byte in the range 1...251 invalid as well (the only valid
+   2-byte code 2 packet is one where the length of both frames is zero).
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|0| N1 (1-2 bytes):                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               :
+     |               Compressed frame 1 (N1 bytes)...                |
+     :                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                               |                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
+     |                     Compressed frame 2...                     :
+     :                                                               |
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                         Figure 4: A Code 2 Packet
+
+3.2.5.  Code 3: A Signaled Number of Frames in the Packet
+
+   Code 3 packets signal the number of frames, as well as additional
+   padding, called "Opus padding" to indicate that this padding is added
+   at the Opus layer rather than at the transport layer.  Code 3 packets
+   MUST have at least 2 bytes [R6,R7].  The TOC byte is followed by a
+   byte encoding the number of frames in the packet in bits 2 to 7
+   (marked "M" in Figure 5), with bit 1 indicating whether or not Opus
+   padding is inserted (marked "p" in Figure 5), and bit 0 indicating
+   VBR (marked "v" in Figure 5).  M MUST NOT be zero, and the audio
+   duration contained within a packet MUST NOT exceed 120 ms [R5].  This
+   limits the maximum frame count for any frame size to 48 (for 2.5 ms
+   frames), with lower limits for longer frame sizes.  Figure 5
+   illustrates the layout of the frame count byte.
+
+                              0
+                              0 1 2 3 4 5 6 7
+                             +-+-+-+-+-+-+-+-+
+                             |v|p|     M     |
+                             +-+-+-+-+-+-+-+-+
+
+                      Figure 5: The frame count byte
+
+   When Opus padding is used, the number of bytes of padding is encoded
+   in the bytes following the frame count byte.  Values from 0...254
+   indicate that 0...254 bytes of padding are included, in addition to
+
+
+
+Valin, et al.                Standards Track                   [Page 18]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   the byte(s) used to indicate the size of the padding.  If the value
+   is 255, then the size of the additional padding is 254 bytes, plus
+   the padding value encoded in the next byte.  There MUST be at least
+   one more byte in the packet in this case [R6,R7].  The additional
+   padding bytes appear at the end of the packet and MUST be set to zero
+   by the encoder to avoid creating a covert channel.  The decoder MUST
+   accept any value for the padding bytes, however.
+
+   Although this encoding provides multiple ways to indicate a given
+   number of padding bytes, each uses a different number of bytes to
+   indicate the padding size and thus will increase the total packet
+   size by a different amount.  For example, to add 255 bytes to a
+   packet, set the padding bit, p, to 1, insert a single byte after the
+   frame count byte with a value of 254, and append 254 padding bytes
+   with the value zero to the end of the packet.  To add 256 bytes to a
+   packet, set the padding bit to 1, insert two bytes after the frame
+   count byte with the values 255 and 0, respectively, and append 254
+   padding bytes with the value zero to the end of the packet.  By using
+   the value 255 multiple times, it is possible to create a packet of
+   any specific, desired size.  Let P be the number of header bytes used
+   to indicate the padding size plus the number of padding bytes
+   themselves (i.e., P is the total number of bytes added to the
+   packet).  Then, P MUST be no more than N-2 [R6,R7].
+
+   In the CBR case, let R=N-2-P be the number of bytes remaining in the
+   packet after subtracting the (optional) padding.  Then, the
+   compressed length of each frame in bytes is equal to R/M.  The value
+   R MUST be a non-negative integer multiple of M [R6].  The compressed
+   data for all M frames follows, each of size R/M bytes, as illustrated
+   in Figure 6.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 19]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|1|0|p|     M     |  Padding length (Optional)    :
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 1 (R/M bytes)...               :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 2 (R/M bytes)...               :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :                              ...                              :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame M (R/M bytes)...               :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :                  Opus Padding (Optional)...                   |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                       Figure 6: A CBR Code 3 Packet
+
+   In the VBR case, the (optional) padding length is followed by M-1
+   frame lengths (indicated by "N1" to "N[M-1]" in Figure 7), each
+   encoded in a one- or two-byte sequence as described above.  The
+   packet MUST contain enough data for the M-1 lengths after removing
+   the (optional) padding, and the sum of these lengths MUST be no
+   larger than the number of bytes remaining in the packet after
+   decoding them [R7].  The compressed data for all M frames follows,
+   each frame consisting of the indicated number of bytes, with the
+   final frame consuming any remaining bytes before the final padding,
+   as illustrated in Figure 6.  The number of header bytes (TOC byte,
+   frame count byte, padding length bytes, and frame length bytes), plus
+   the signaled length of the first M-1 frames themselves, plus the
+   signaled length of the padding MUST be no larger than N, the total
+   size of the packet.
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 20]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|1|1|p|     M     | Padding length (Optional)     :
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     : N1 (1-2 bytes): N2 (1-2 bytes):     ...       :     N[M-1]    |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 1 (N1 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 2 (N2 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :                              ...                              :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :                     Compressed frame M...                     :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :                  Opus Padding (Optional)...                   |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                       Figure 7: A VBR Code 3 Packet
+
+3.3.  Examples
+
+   Simplest case, one NB mono 20 ms SILK frame:
+
+    0                   1                   2                   3
+    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   |    1    |0|0|0|               compressed data...              :
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                                 Figure 8
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 21]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Two FB mono 5 ms CELT frames of the same compressed size:
+
+    0                   1                   2                   3
+    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   |   29    |0|0|1|               compressed data...              :
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                                 Figure 9
+
+   Two FB mono 20 ms Hybrid frames of different compressed size:
+
+    0                   1                   2                   3
+    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   |   15    |0|1|1|1|0|     2     |      N1       |               |
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               |
+   |                       compressed data...                      :
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                                 Figure 10
+
+   Four FB stereo 20 ms CELT frames of the same compressed size:
+
+    0                   1                   2                   3
+    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+   |   31    |1|1|1|0|0|     4     |      compressed data...       :
+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                                 Figure 11
+
+3.4.  Receiving Malformed Packets
+
+   A receiver MUST NOT process packets that violate any of the rules
+   above as normal Opus packets.  They are reserved for future
+   applications, such as in-band headers (containing metadata, etc.).
+   Packets that violate these constraints may cause implementations of
+   _this_ specification to treat them as malformed and discard them.
+
+   These constraints are summarized here for reference:
+
+   [R1]  Packets are at least one byte.
+
+   [R2]  No implicit frame length is larger than 1275 bytes.
+
+   [R3]  Code 1 packets have an odd total length, N, so that (N-1)/2 is
+         an integer.
+
+
+
+Valin, et al.                Standards Track                   [Page 22]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   [R4]  Code 2 packets have enough bytes after the TOC for a valid
+         frame length, and that length is no larger than the number of
+         bytes remaining in the packet.
+
+   [R5]  Code 3 packets contain at least one frame, but no more than
+         120 ms of audio total.
+
+   [R6]  The length of a CBR code 3 packet, N, is at least two bytes,
+         the number of bytes added to indicate the padding size plus the
+         trailing padding bytes themselves, P, is no more than N-2, and
+         the frame count, M, satisfies the constraint that (N-2-P) is a
+         non-negative integer multiple of M.
+
+   [R7]  VBR code 3 packets are large enough to contain all the header
+         bytes (TOC byte, frame count byte, any padding length bytes,
+         and any frame length bytes), plus the length of the first M-1
+         frames, plus any trailing padding bytes.
+
+4.  Opus Decoder
+
+   The Opus decoder consists of two main blocks: the SILK decoder and
+   the CELT decoder.  At any given time, one or both of the SILK and
+   CELT decoders may be active.  The output of the Opus decode is the
+   sum of the outputs from the SILK and CELT decoders with proper sample
+   rate conversion and delay compensation on the SILK side, and optional
+   decimation (when decoding to sample rates less than 48 kHz) on the
+   CELT side, as illustrated in the block diagram below.
+
+
+                            +---------+    +------------+
+                            |  SILK   |    |   Sample   |
+                         +->| Decoder |--->|    Rate    |----+
+   Bit-    +---------+   |  |         |    | Conversion |    v
+   stream  |  Range  |---+  +---------+    +------------+  /---\  Audio
+   ------->| Decoder |                                     | + |------>
+           |         |---+  +---------+    +------------+  \---/
+           +---------+   |  |  CELT   |    | Decimation |    ^
+                         +->| Decoder |--->| (Optional) |----+
+                            |         |    |            |
+                            +---------+    +------------+
+
+
+4.1.  Range Decoder
+
+   Opus uses an entropy coder based on range coding [RANGE-CODING]
+   [MARTIN79], which is itself a rediscovery of the FIFO arithmetic code
+   introduced by [CODING-THESIS].  It is very similar to arithmetic
+   encoding, except that encoding is done with digits in any base
+
+
+
+Valin, et al.                Standards Track                   [Page 23]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   instead of with bits, so it is faster when using larger bases (i.e.,
+   a byte).  All of the calculations in the range coder must use bit-
+   exact integer arithmetic.
+
+   Symbols may also be coded as "raw bits" packed directly into the
+   bitstream, bypassing the range coder.  These are packed backwards
+   starting at the end of the frame, as illustrated in Figure 12.  This
+   reduces complexity and makes the stream more resilient to bit errors,
+   as corruption in the raw bits will not desynchronize the decoding
+   process, unlike corruption in the input to the range decoder.  Raw
+   bits are only used in the CELT layer.
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | Range coder data (packed MSB to LSB) ->                       :
+     +                                                               +
+     :                                                               :
+     +     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :     | <- Boundary occurs at an arbitrary bit position         :
+     +-+-+-+                                                         +
+     :                          <- Raw bits data (packed LSB to MSB) |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+     Legend:
+
+     LSB = Least Significant Bit
+     MSB = Most Significant Bit
+
+          Figure 12: Illustrative Example of Packing Range Coder
+                             and Raw Bits Data
+
+   Each symbol coded by the range coder is drawn from a finite alphabet
+   and coded in a separate "context", which describes the size of the
+   alphabet and the relative frequency of each symbol in that alphabet.
+
+   Suppose there is a context with n symbols, identified with an index
+   that ranges from 0 to n-1.  The parameters needed to encode or decode
+   symbol k in this context are represented by a three-tuple
+   (fl[k], fh[k], ft), all 16-bit unsigned integers, with
+   0 <= fl[k] < fh[k] <= ft <= 65535.  The values of this tuple are
+   derived from the probability model for the symbol, represented by
+   traditional "frequency counts".  Because Opus uses static contexts,
+   those are not updated as symbols are decoded.  Let f[i] be the
+   frequency of symbol i.  Then, the three-tuple corresponding to symbol
+   k is given by the following:
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 24]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                   k-1                                   n-1
+                   __                                    __
+           fl[k] = \  f[i],  fh[k] = fl[k] + f[k],  ft = \  f[i]
+                   /_                                    /_
+                   i=0                                   i=0
+
+   The range decoder extracts the symbols and integers encoded using the
+   range encoder in Section 5.1.  The range decoder maintains an
+   internal state vector composed of the two-tuple (val, rng), where val
+   represents the difference between the high end of the current range
+   and the actual coded value, minus one, and rng represents the size of
+   the current range.  Both val and rng are 32-bit unsigned integer
+   values.
+
+4.1.1.  Range Decoder Initialization
+
+   Let b0 be an 8-bit unsigned integer containing first input byte (or
+   containing zero if there are no bytes in this Opus frame).  The
+   decoder initializes rng to 128 and initializes val to (127 -
+    (b0>>1)), where (b0>>1) is the top 7 bits of the first input byte.
+   It saves the remaining bit, (b0&1), for use in the renormalization
+   procedure described in Section 4.1.2.1, which the decoder invokes
+   immediately after initialization to read additional bits and
+   establish the invariant that rng > 2**23.
+
+4.1.2.  Decoding Symbols
+
+   Decoding a symbol is a two-step process.  The first step determines a
+   16-bit unsigned value fs, which lies within the range of some symbol
+   in the current context.  The second step updates the range decoder
+   state with the three-tuple (fl[k], fh[k], ft) corresponding to that
+   symbol.
+
+   The first step is implemented by ec_decode() (entdec.c), which
+   computes
+
+                                      val
+                       fs = ft - min(------ + 1, ft)
+                                     rng/ft
+
+   The divisions here are integer division.
+
+   The decoder then identifies the symbol in the current context
+   corresponding to fs; i.e., the value of k whose three-tuple
+   (fl[k], fh[k], ft) satisfies fl[k] <= fs < fh[k].  It uses this tuple
+   to update val according to
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 25]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                                   rng
+                       val = val - --- * (ft - fh[k])
+                                   ft
+
+   If fl[k] is greater than zero, then the decoder updates rng using
+
+                              rng
+                        rng = --- * (fh[k] - fl[k])
+                              ft
+
+   Otherwise, it updates rng using
+
+                                   rng
+                       rng = rng - --- * (ft - fh[k])
+                                   ft
+
+   Using a special case for the first symbol (rather than the last
+   symbol, as is commonly done in other arithmetic coders) ensures that
+   all the truncation error from the finite precision arithmetic
+   accumulates in symbol 0.  This makes the cost of coding a 0 slightly
+   smaller, on average, than its estimated probability indicates and
+   makes the cost of coding any other symbol slightly larger.  When
+   contexts are designed so that 0 is the most probable symbol, which is
+   often the case, this strategy minimizes the inefficiency introduced
+   by the finite precision.  It also makes some of the special-case
+   decoding routines in Section 4.1.3 particularly simple.
+
+   After the updates, implemented by ec_dec_update() (entdec.c), the
+   decoder normalizes the range using the procedure in the next section,
+   and returns the index k.
+
+4.1.2.1.  Renormalization
+
+   To normalize the range, the decoder repeats the following process,
+   implemented by ec_dec_normalize() (entdec.c), until rng > 2**23.  If
+   rng is already greater than 2**23, the entire process is skipped.
+   First, it sets rng to (rng<<8).  Then, it reads the next byte of the
+   Opus frame and forms an 8-bit value sym, using the leftover bit
+   buffered from the previous byte as the high bit and the top 7 bits of
+   the byte just read as the other 7 bits of sym.  The remaining bit in
+   the byte just read is buffered for use in the next iteration.  If no
+   more input bytes remain, it uses zero bits instead.  See
+   Section 4.1.1 for the initialization used to process the first byte.
+   Then, it sets
+
+                 val = ((val<<8) + (255-sym)) & 0x7FFFFFFF
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 26]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   It is normal and expected that the range decoder will read several
+   bytes into the data of the raw bits (if any) at the end of the frame
+   by the time the frame is completely decoded, as illustrated in
+   Figure 13.  This same data MUST also be returned as raw bits when
+   requested.  The encoder is expected to terminate the stream in such a
+   way that the range decoder will decode the intended values regardless
+   of the data contained in the raw bits.  Section 5.1.5 describes a
+   procedure for doing this.  If the range decoder consumes all of the
+   bytes belonging to the current frame, it MUST continue to use zero
+   when any further input bytes are required, even if there is
+   additional data in the current packet from padding or other frames.
+
+      n              n+1             n+2             n+3
+      0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :     | <----------- Overlap region ------------> |             :
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+           ^                                           ^
+           |   End of data buffered by the range coder |
+     ...-----------------------------------------------+
+           |
+           | End of data consumed by raw bits
+           +-------------------------------------------------------...
+
+          Figure 13: Illustrative Example of Raw Bits Overlapping
+                             Range Coder Data
+
+4.1.3.  Alternate Decoding Methods
+
+   The reference implementation uses three additional decoding methods
+   that are exactly equivalent to the above but make assumptions and
+   simplifications that allow for a more efficient implementation.
+
+4.1.3.1.  ec_decode_bin()
+
+   The first is ec_decode_bin() (entdec.c), defined using the parameter
+   ftb instead of ft.  It is mathematically equivalent to calling
+   ec_decode() with ft = (1<<ftb), but it avoids one of the divisions.
+
+4.1.3.2.  ec_dec_bit_logp()
+
+   The next is ec_dec_bit_logp() (entdec.c), which decodes a single
+   binary symbol, replacing both the ec_decode() and ec_dec_update()
+   steps.  The context is described by a single parameter, logp, which
+   is the absolute value of the base-2 logarithm of the probability of a
+   "1".  It is mathematically equivalent to calling ec_decode() with
+   ft = (1<<logp), followed by ec_dec_update() with the 3-tuple
+   (fl[k] = 0, fh[k] = (1<<logp) - 1, ft = (1<<logp)) if the returned
+
+
+
+Valin, et al.                Standards Track                   [Page 27]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   value of fs is less than (1<<logp) - 1 (a "0" was decoded), and with
+   (fl[k] = (1<<logp) - 1, fh[k] = ft = (1<<logp)) otherwise (a "1" was
+   decoded).  The implementation requires no multiplications or
+   divisions.
+
+4.1.3.3.  ec_dec_icdf()
+
+   The last is ec_dec_icdf() (entdec.c), which decodes a single symbol
+   with a table-based context of up to 8 bits, also replacing both the
+   ec_decode() and ec_dec_update() steps, as well as the search for the
+   decoded symbol in between.  The context is described by two
+   parameters, an icdf ("inverse" cumulative distribution function)
+   table and ftb.  As with ec_decode_bin(), (1<<ftb) is equivalent to
+   ft. idcf[k], on the other hand, stores (1<<ftb)-fh[k], which is equal
+   to (1<<ftb) - fl[k+1]. fl[0] is assumed to be 0, and the table is
+   terminated by a value of 0 (where fh[k] == ft).
+
+   The function is mathematically equivalent to calling ec_decode() with
+   ft = (1<<ftb), using the returned value fs to search the table for
+   the first entry where fs < (1<<ftb)-icdf[k], and calling
+   ec_dec_update() with fl[k] = (1<<ftb) - icdf[k-1] (or 0 if k == 0),
+   fh[k] = (1<<ftb) - idcf[k], and ft = (1<<ftb).  Combining the search
+   with the update allows the division to be replaced by a series of
+   multiplications (which are usually much cheaper), and using an
+   inverse CDF allows the use of an ftb as large as 8 in an 8-bit table
+   without any special cases.  This is the primary interface with the
+   range decoder in the SILK layer, though it is used in a few places in
+   the CELT layer as well.
+
+   Although icdf[k] is more convenient for the code, the frequency
+   counts, f[k], are a more natural representation of the probability
+   distribution function (PDF) for a given symbol.  Therefore, this
+   document lists the latter, not the former, when describing the
+   context in which a symbol is coded as a list, e.g., {4, 4, 4, 4}/16
+   for a uniform context with four possible values and ft = 16.  The
+   value of ft after the slash is always the sum of the entries in the
+   PDF, but is included for convenience.  Contexts with identical
+   probabilities, f[k]/ft, but different values of ft (or equivalently,
+   ftb) are not the same, and cannot, in general, be used in place of
+   one another.  An icdf table is also not capable of representing a PDF
+   where the first symbol has 0 probability.  In such contexts,
+   ec_dec_icdf() can decode the symbol by using a table that drops the
+   entries for any initial zero-probability values and by adding the
+   constant offset of the first value with a non-zero probability to its
+   return value.
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 28]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.1.4.  Decoding Raw Bits
+
+   The raw bits used by the CELT layer are packed at the end of the
+   frame, with the least significant bit of the first value packed in
+   the least significant bit of the last byte, filling up to the most
+   significant bit in the last byte, continuing on to the least
+   significant bit of the penultimate byte, and so on.  The reference
+   implementation reads them using ec_dec_bits() (entdec.c).  Because
+   the range decoder must read several bytes ahead in the stream, as
+   described in Section 4.1.2.1, the input consumed by the raw bits may
+   overlap with the input consumed by the range coder, and a decoder
+   MUST allow this.  The format should render it impossible to attempt
+   to read more raw bits than there are actual bits in the frame, though
+   a decoder may wish to check for this and report an error.
+
+4.1.5.  Decoding Uniformly Distributed Integers
+
+   The function ec_dec_uint() (entdec.c) decodes one of ft equiprobable
+   values in the range 0 to (ft - 1), inclusive, each with a frequency
+   of 1, where ft may be as large as (2**32 - 1).  Because ec_decode()
+   is limited to a total frequency of (2**16 - 1), 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.  The limit of 8 bits in the range coded symbol is a trade-off
+   between implementation complexity, modeling error (since the symbols
+   no longer truly have equal coding cost), and rounding error
+   introduced by the range coder itself (which gets larger as more bits
+   are included).  Using raw bits reduces the maximum number of
+   divisions required in the worst case, but means that it may be
+   possible to decode a value outside the range 0 to (ft - 1),
+   inclusive.
+
+   ec_dec_uint() takes a single, positive parameter, ft, which is not
+   necessarily a power of two, and returns an integer, t, whose value
+   lies between 0 and (ft - 1), inclusive.  Let ftb = ilog(ft - 1),
+   i.e., the number of bits required to store (ft - 1) in two's
+   complement notation.  If ftb is 8 or less, then t is decoded with
+   t = ec_decode(ft), and the range coder state is updated using the
+   three-tuple (t, t + 1, ft).
+
+   If ftb is greater than 8, then the top 8 bits of t are decoded using
+
+                 t = ec_decode(((ft - 1) >> (ftb - 8)) + 1)
+
+   the decoder state is updated using the three-tuple (t, t + 1, ((ft -
+    1) >> (ftb - 8)) + 1), and the remaining bits are decoded as raw
+   bits, setting
+
+
+
+
+Valin, et al.                Standards Track                   [Page 29]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                t = (t << (ftb - 8)) | ec_dec_bits(ftb - 8)
+
+   If, at this point, t >= ft, then the current frame is corrupt.  In
+   that case, the decoder should assume there has been an error in the
+   coding, decoding, or transmission and SHOULD take measures to conceal
+   the error (e.g., saturate to ft-1 or use the Packet Loss Concealment
+   (PLC)) and/or report to the application that the error has occurred.
+
+4.1.6.  Current Bit Usage
+
+   The bit allocation routines in the CELT decoder need a conservative
+   upper bound on the number of bits that have been used from the
+   current frame thus far, including both range coder bits and raw bits.
+   This drives allocation decisions that must match those made in the
+   encoder.  The upper bound is computed in the reference implementation
+   to whole-bit precision by the function ec_tell() (entcode.h) and to
+   fractional 1/8th bit precision by the function ec_tell_frac()
+   (entcode.c).  Like all operations in the range coder, it must be
+   implemented in a bit-exact manner, and it must produce exactly the
+   same value returned by the same functions in the encoder after
+   encoding the same symbols.
+
+   ec_tell() is guaranteed to return ceil(ec_tell_frac()/8.0).  In
+   various places, the codec will check to ensure there is enough room
+   to contain a symbol before attempting to decode it.  In practice,
+   although the number of bits used so far is an upper bound, decoding a
+   symbol whose probability model suggests it has a worst-case cost of p
+   1/8th bits may actually advance the return value of ec_tell_frac() by
+   p-1, p, or p+1 1/8th bits, due to approximation error in that upper
+   bound, truncation error in the range coder, and for large values of
+   ft, modeling error in ec_dec_uint().
+
+   However, this error is bounded, and periodic calls to ec_tell() or
+   ec_tell_frac() at precisely defined points in the decoding process
+   prevent it from accumulating.  For a range coder symbol that requires
+   a whole number of bits (i.e., for which ft/(fh[k] - fl[k]) is a power
+   of two), where there are at least p 1/8th bits available, decoding
+   the symbol will never cause ec_tell() or ec_tell_frac() to exceed the
+   size of the frame ("bust the budget").  In this case, the return
+   value of ec_tell_frac() will only advance by more than p 1/8th bits
+   if there were an additional, fractional number of bits remaining, and
+   it will never advance beyond the next whole-bit boundary, which is
+   safe, since frames always contain a whole number of bits.  However,
+   when p is not a whole number of bits, an extra 1/8th bit is required
+   to ensure that decoding the symbol will not bust the budget.
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 30]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The reference implementation keeps track of the total number of whole
+   bits that have been processed by the decoder so far in the variable
+   nbits_total, including the (possibly fractional) number of bits that
+   are currently buffered, but not consumed, inside the range coder.
+   nbits_total is initialized to 9 just before the initial range
+   renormalization process completes (or equivalently, it can be
+   initialized to 33 after the first renormalization).  The extra two
+   bits over the actual amount buffered by the range coder guarantees
+   that it is an upper bound and that there is enough room for the
+   encoder to terminate the stream.  Each iteration through the range
+   coder's renormalization loop increases nbits_total by 8.  Reading raw
+   bits increases nbits_total by the number of raw bits read.
+
+4.1.6.1.  ec_tell()
+
+   The whole number of bits buffered in rng may be estimated via
+   lg = ilog(rng). ec_tell() then becomes a simple matter of removing
+   these bits from the total.  It returns (nbits_total - lg).
+
+   In a newly initialized decoder, before any symbols have been read,
+   this reports that 1 bit has been used.  This is the bit reserved for
+   termination of the encoder.
+
+4.1.6.2.  ec_tell_frac()
+
+   ec_tell_frac() estimates the number of bits buffered in rng to
+   fractional precision.  Since rng must be greater than 2**23 after
+   renormalization, lg must be at least 24.  Let
+
+                           r_Q15 = rng >> (lg-16)
+
+   so that 32768 <= r_Q15 < 65536, an unsigned Q15 value representing
+   the fractional part of rng.  Then, the following procedure can be
+   used to add one bit of precision to lg.  First, update
+
+                        r_Q15 = (r_Q15*r_Q15) >> 15
+
+   Then, add the 16th bit of r_Q15 to lg via
+
+                         lg = 2*lg + (r_Q15 >> 16)
+
+   Finally, if this bit was a 1, reduce r_Q15 by a factor of two via
+
+                             r_Q15 = r_Q15 >> 1
+
+   so that it once again lies in the range 32768 <= r_Q15 < 65536.  This
+   procedure is repeated three times to extend lg to 1/8th bit
+   precision. ec_tell_frac() then returns (nbits_total*8 - lg).
+
+
+
+Valin, et al.                Standards Track                   [Page 31]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.2.  SILK Decoder
+
+   The decoder's LP layer uses a modified version of the SILK codec
+   (herein simply called "SILK"), which runs a decoded excitation signal
+   through adaptive long-term and short-term prediction synthesis
+   filters.  It runs at NB, MB, and WB sample rates internally.  When
+   used in a SWB or FB Hybrid frame, the LP layer itself still only runs
+   in WB.
+
+4.2.1.  SILK Decoder Modules
+
+   An overview of the decoder is given in Figure 14.
+
+        +---------+    +------------+
+     -->| Range   |--->| Decode     |---------------------------+
+      1 | Decoder | 2  | Parameters |----------+       5        |
+        +---------+    +------------+     4    |                |
+                            3 |                |                |
+                             \/               \/               \/
+                       +------------+   +------------+   +------------+
+                       | Generate   |-->| LTP        |-->| LPC        |
+                       | Excitation |   | Synthesis  |   | Synthesis  |
+                       +------------+   +------------+   +------------+
+                                               ^                |
+                                               |                |
+                           +-------------------+----------------+
+                           |                                      6
+                           |   +------------+   +-------------+
+                           +-->| Stereo     |-->| Sample Rate |-->
+                               | Unmixing   | 7 | Conversion  | 8
+                               +------------+   +-------------+
+
+     1: Range encoded bitstream
+     2: Coded parameters
+     3: Pulses, LSBs, and signs
+     4: Pitch lags, Long-Term Prediction (LTP) coefficients
+     5: Linear Predictive Coding (LPC) coefficients and gains
+     6: Decoded signal (mono or mid-side stereo)
+     7: Unmixed signal (mono or left-right stereo)
+     8: Resampled signal
+
+
+                          Figure 14: SILK Decoder
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 32]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The decoder feeds the bitstream (1) to the range decoder from
+   Section 4.1 and then decodes the parameters in it (2) using the
+   procedures detailed in Sections 4.2.3 through 4.2.7.8.5.  These
+   parameters (3, 4, 5) are used to generate an excitation signal (see
+   Section 4.2.7.8.6), which is fed to an optional Long-Term Prediction
+   (LTP) filter (voiced frames only, see Section 4.2.7.9.1) and then a
+   short-term prediction filter (see Section 4.2.7.9.2), producing the
+   decoded signal (6).  For stereo streams, the mid-side representation
+   is converted to separate left and right channels (7).  The result is
+   finally resampled to the desired output sample rate (e.g., 48 kHz) so
+   that the resampled signal (8) can be mixed with the CELT layer.
+
+4.2.2.  LP Layer Organization
+
+   Internally, the LP layer of a single Opus frame is composed of either
+   a single 10 ms regular SILK frame or between one and three 20 ms
+   regular SILK frames.  A stereo Opus frame may double the number of
+   regular SILK frames (up to a total of six), since it includes
+   separate frames for a mid channel and, optionally, a side channel.
+   Optional Low Bit-Rate Redundancy (LBRR) frames, which are reduced-
+   bitrate encodings of previous SILK frames, may be included to aid in
+   recovery from packet loss.  If present, these appear before the
+   regular SILK frames.  They are, in most respects, identical to
+   regular, active SILK frames, except that they are usually encoded
+   with a lower bitrate.  This document uses "SILK frame" to refer to
+   either one and "regular SILK frame" if it needs to draw a distinction
+   between the two.
+
+   Logically, each SILK frame is, in turn, composed of either two or
+   four 5 ms subframes.  Various parameters, such as the quantization
+   gain of the excitation and the pitch lag and filter coefficients can
+   vary on a subframe-by-subframe basis.  Physically, the parameters for
+   each subframe are interleaved in the bitstream, as described in the
+   relevant sections for each parameter.
+
+   All of these frames and subframes are decoded from the same range
+   coder, with no padding between them.  Thus, packing multiple SILK
+   frames in a single Opus frame saves, on average, half a byte per SILK
+   frame.  It also allows some parameters to be predicted from prior
+   SILK frames in the same Opus frame, since this does not degrade
+   packet loss robustness (beyond any penalty for merely using fewer,
+   larger packets to store multiple frames).
+
+   Stereo support in SILK uses a variant of mid-side coding, allowing a
+   mono decoder to simply decode the mid channel.  However, the data for
+   the two channels is interleaved, so a mono decoder must still unpack
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 33]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   the data for the side channel.  It would be required to do so anyway
+   for Hybrid Opus frames or to support decoding individual 20 ms
+   frames.
+
+   Table 3 summarizes the overall grouping of the contents of the LP
+   layer.  Figures 15 and 16 illustrate the ordering of the various SILK
+   frames for a 60 ms Opus frame, for both mono and stereo,
+   respectively.
+
+   +-----------------------------------+---------------+---------------+
+   |             Symbol(s)             |     PDF(s)    |   Condition   |
+   +-----------------------------------+---------------+---------------+
+   |   Voice Activity Detection (VAD)  |    {1, 1}/2   |               |
+   |               Flags               |               |               |
+   |                                   |               |               |
+   |             LBRR Flag             |    {1, 1}/2   |               |
+   |                                   |               |               |
+   |        Per-Frame LBRR Flags       |    Table 4    | Section 4.2.4 |
+   |                                   |               |               |
+   |           LBRR Frame(s)           | Section 4.2.7 | Section 4.2.4 |
+   |                                   |               |               |
+   |       Regular SILK Frame(s)       | Section 4.2.7 |               |
+   +-----------------------------------+---------------+---------------+
+
+         Table 3: Organization of the SILK layer of an Opus Frame
+
+
+                    +---------------------------------+
+                    |            VAD Flags            |
+                    +---------------------------------+
+                    |            LBRR Flag            |
+                    +---------------------------------+
+                    | Per-Frame LBRR Flags (Optional) |
+                    +---------------------------------+
+                    |     LBRR Frame 1 (Optional)     |
+                    +---------------------------------+
+                    |     LBRR Frame 2 (Optional)     |
+                    +---------------------------------+
+                    |     LBRR Frame 3 (Optional)     |
+                    +---------------------------------+
+                    |      Regular SILK Frame 1       |
+                    +---------------------------------+
+                    |      Regular SILK Frame 2       |
+                    +---------------------------------+
+                    |      Regular SILK Frame 3       |
+                    +---------------------------------+
+
+                       Figure 15: A 60 ms Mono Frame
+
+
+
+Valin, et al.                Standards Track                   [Page 34]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                 +---------------------------------------+
+                 |             Mid VAD Flags             |
+                 +---------------------------------------+
+                 |             Mid LBRR Flag             |
+                 +---------------------------------------+
+                 |             Side VAD Flags            |
+                 +---------------------------------------+
+                 |             Side LBRR Flag            |
+                 +---------------------------------------+
+                 |  Mid Per-Frame LBRR Flags (Optional)  |
+                 +---------------------------------------+
+                 | Side Per-Frame LBRR Flags (Optional)  |
+                 +---------------------------------------+
+                 |     Mid LBRR Frame 1 (Optional)       |
+                 +---------------------------------------+
+                 |     Side LBRR Frame 1 (Optional)      |
+                 +---------------------------------------+
+                 |     Mid LBRR Frame 2 (Optional)       |
+                 +---------------------------------------+
+                 |     Side LBRR Frame 2 (Optional)      |
+                 +---------------------------------------+
+                 |     Mid LBRR Frame 3 (Optional)       |
+                 +---------------------------------------+
+                 |     Side LBRR Frame 3 (Optional)      |
+                 +---------------------------------------+
+                 |      Mid Regular SILK Frame 1         |
+                 +---------------------------------------+
+                 | Side Regular SILK Frame 1 (Optional)  |
+                 +---------------------------------------+
+                 |      Mid Regular SILK Frame 2         |
+                 +---------------------------------------+
+                 | Side Regular SILK Frame 2 (Optional)  |
+                 +---------------------------------------+
+                 |      Mid Regular SILK Frame 3         |
+                 +---------------------------------------+
+                 | Side Regular SILK Frame 3 (Optional)  |
+                 +---------------------------------------+
+
+                      Figure 16: A 60 ms Stereo Frame
+
+4.2.3.  Header Bits
+
+   The LP layer begins with two to eight header bits, decoded in
+   silk_Decode() (dec_API.c).  These consist of one Voice Activity
+   Detection (VAD) bit per frame (up to 3), followed by a single flag
+   indicating the presence of LBRR frames.  For a stereo packet, these
+   first flags correspond to the mid channel, and a second set of flags
+   is included for the side channel.
+
+
+
+Valin, et al.                Standards Track                   [Page 35]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Because these are the first symbols decoded by the range coder and
+   because they are coded as binary values with uniform probability,
+   they can be extracted directly from the most significant bits of the
+   first byte of compressed data.  Thus, a receiver can determine if an
+   Opus frame contains any active SILK frames without the overhead of
+   using the range decoder.
+
+4.2.4.  Per-Frame LBRR Flags
+
+   For Opus frames longer than 20 ms, a set of LBRR flags is decoded for
+   each channel that has its LBRR flag set.  Each set contains one flag
+   per 20 ms SILK frame. 40 ms Opus frames use the 2-frame LBRR flag PDF
+   from Table 4, and 60 ms Opus frames use the 3-frame LBRR flag PDF.
+   For each channel, the resulting 2- or 3-bit integer contains the
+   corresponding LBRR flag for each frame, packed in order from the LSB
+   to the MSB.
+
+           +------------+-------------------------------------+
+           | Frame Size | PDF                                 |
+           +------------+-------------------------------------+
+           | 40 ms      | {0, 53, 53, 150}/256                |
+           |            |                                     |
+           | 60 ms      | {0, 41, 20, 29, 41, 15, 28, 82}/256 |
+           +------------+-------------------------------------+
+
+                          Table 4: LBRR Flag PDFs
+
+   A 10 or 20 ms Opus frame does not contain any per-frame LBRR flags,
+   as there may be at most one LBRR frame per channel.  The global LBRR
+   flag in the header bits (see Section 4.2.3) is already sufficient to
+   indicate the presence of that single LBRR frame.
+
+4.2.5.  LBRR Frames
+
+   The LBRR frames, if present, contain an encoded representation of the
+   signal immediately prior to the current Opus frame as if it were
+   encoded with the current mode, frame size, audio bandwidth, and
+   channel count, even if those differ from the prior Opus frame.  When
+   one of these parameters changes from one Opus frame to the next, this
+   implies that the LBRR frames of the current Opus frame may not be
+   simple drop-in replacements for the contents of the previous Opus
+   frame.
+
+   For example, when switching from 20 ms to 60 ms, the 60 ms Opus frame
+   may contain LBRR frames covering up to three prior 20 ms Opus frames,
+   even if those frames already contained LBRR frames covering some of
+   the same time periods.  When switching from 20 ms to 10 ms, the 10 ms
+   Opus frame can contain an LBRR frame covering at most half the prior
+
+
+
+Valin, et al.                Standards Track                   [Page 36]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   20 ms Opus frame, potentially leaving a hole that needs to be
+   concealed from even a single packet loss (see Section 4.4).  When
+   switching from mono to stereo, the LBRR frames in the first stereo
+   Opus frame MAY contain a non-trivial side channel.
+
+   In order to properly produce LBRR frames under all conditions, an
+   encoder might need to buffer up to 60 ms of audio and re-encode it
+   during these transitions.  However, the reference implementation opts
+   to disable LBRR frames at the transition point for simplicity.  Since
+   transitions are relatively infrequent in normal usage, this does not
+   have a significant impact on packet loss robustness.
+
+   The LBRR frames immediately follow the LBRR flags, prior to any
+   regular SILK frames.  Section 4.2.7 describes their exact contents.
+   LBRR frames do not include their own separate VAD flags.  LBRR frames
+   are only meant to be transmitted for active speech, thus all LBRR
+   frames are treated as active.
+
+   In a stereo Opus frame longer than 20 ms, although the per-frame LBRR
+   flags for the mid channel are coded as a unit before the per-frame
+   LBRR flags for the side channel, the LBRR frames themselves are
+   interleaved.  The decoder parses an LBRR frame for the mid channel of
+   a given 20 ms interval (if present) and then immediately parses the
+   corresponding LBRR frame for the side channel (if present), before
+   proceeding to the next 20 ms interval.
+
+4.2.6.  Regular SILK Frames
+
+   The regular SILK frame(s) follow the LBRR frames (if any).
+   Section 4.2.7 describes their contents, as well.  Unlike the LBRR
+   frames, a regular SILK frame is coded for each time interval in an
+   Opus frame, even if the corresponding VAD flags are unset.  For
+   stereo Opus frames longer than 20 ms, the regular mid and side SILK
+   frames for each 20 ms interval are interleaved, just as with the LBRR
+   frames.  The side frame may be skipped by coding an appropriate flag,
+   as detailed in Section 4.2.7.2.
+
+4.2.7.  SILK Frame Contents
+
+   Each SILK frame includes a set of side information that encodes
+
+   o  The frame type and quantization type (Section 4.2.7.3),
+
+   o  Quantization gains (Section 4.2.7.4),
+
+   o  Short-term prediction filter coefficients (Section 4.2.7.5),
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 37]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  A Line Spectral Frequencies (LSFs) interpolation weight
+      (Section 4.2.7.5.5),
+
+   o  LTP filter lags and gains (Section 4.2.7.6), and
+
+   o  A Linear Congruential Generator (LCG) seed (Section 4.2.7.7).
+
+   The quantized excitation signal (see Section 4.2.7.8) follows these
+   at the end of the frame.  Table 5 details the overall organization of
+   a SILK frame.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 38]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +---------------------------+-------------------+-------------------+
+   |         Symbol(s)         |       PDF(s)      |     Condition     |
+   +---------------------------+-------------------+-------------------+
+   | Stereo Prediction Weights |      Table 6      |  Section 4.2.7.1  |
+   |                           |                   |                   |
+   |       Mid-only Flag       |      Table 8      |  Section 4.2.7.2  |
+   |                           |                   |                   |
+   |         Frame Type        |  Section 4.2.7.3  |                   |
+   |                           |                   |                   |
+   |       Subframe Gains      |  Section 4.2.7.4  |                   |
+   |                           |                   |                   |
+   |   Normalized LSF Stage-1  |      Table 14     |                   |
+   |           Index           |                   |                   |
+   |                           |                   |                   |
+   |   Normalized LSF Stage-2  | Section 4.2.7.5.2 |                   |
+   |          Residual         |                   |                   |
+   |                           |                   |                   |
+   |       Normalized LSF      |      Table 26     |    20 ms frame    |
+   |    Interpolation Weight   |                   |                   |
+   |                           |                   |                   |
+   |     Primary Pitch Lag     | Section 4.2.7.6.1 |    Voiced frame   |
+   |                           |                   |                   |
+   |   Subframe Pitch Contour  |      Table 32     |    Voiced frame   |
+   |                           |                   |                   |
+   |     Periodicity Index     |      Table 37     |    Voiced frame   |
+   |                           |                   |                   |
+   |         LTP Filter        |      Table 38     |    Voiced frame   |
+   |                           |                   |                   |
+   |        LTP Scaling        |      Table 42     | Section 4.2.7.6.3 |
+   |                           |                   |                   |
+   |          LCG Seed         |      Table 43     |                   |
+   |                           |                   |                   |
+   |   Excitation Rate Level   |      Table 45     |                   |
+   |                           |                   |                   |
+   |  Excitation Pulse Counts  |      Table 46     |                   |
+   |                           |                   |                   |
+   |      Excitation Pulse     | Section 4.2.7.8.3 |   Non-zero pulse  |
+   |         Locations         |                   |       count       |
+   |                           |                   |                   |
+   |      Excitation LSBs      |      Table 51     | Section 4.2.7.8.2 |
+   |                           |                   |                   |
+   |      Excitation Signs     |      Table 52     |                   |
+   +---------------------------+-------------------+-------------------+
+
+         Table 5: Order of the Symbols in an Individual SILK Frame
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 39]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.2.7.1.  Stereo Prediction Weights
+
+   A SILK frame corresponding to the mid channel of a stereo Opus frame
+   begins with a pair of side channel prediction weights, designed such
+   that zeros indicate normal mid-side coupling.  Since these weights
+   can change on every frame, the first portion of each frame linearly
+   interpolates between the previous weights and the current ones, using
+   zeros for the previous weights if none are available.  These
+   prediction weights are never included in a mono Opus frame, and the
+   previous weights are reset to zeros on any transition from mono to
+   stereo.  They are also not included in an LBRR frame for the side
+   channel, even if the LBRR flags indicate the corresponding mid
+   channel was not coded.  In that case, the previous weights are used,
+   again substituting in zeros if no previous weights are available
+   since the last decoder reset (see Section 4.5.2).
+
+   To summarize, these weights are coded if and only if
+
+   o  This is a stereo Opus frame (Section 3.1), and
+
+   o  The current SILK frame corresponds to the mid channel.
+
+   The prediction weights are coded in three separate pieces, which are
+   decoded by silk_stereo_decode_pred() (stereo_decode_pred.c).  The
+   first piece jointly codes the high-order part of a table index for
+   both weights.  The second piece codes the low-order part of each
+   table index.  The third piece codes an offset used to linearly
+   interpolate between table indices.  The details are as follows.
+
+   Let n be an index decoded with the 25-element stage-1 PDF in Table 6.
+   Then, let i0 and i1 be indices decoded with the stage-2 and stage-3
+   PDFs in Table 6, respectively, and let i2 and i3 be two more indices
+   decoded with the stage-2 and stage-3 PDFs, all in that order.
+
+   +-------+-----------------------------------------------------------+
+   | Stage | PDF                                                       |
+   +-------+-----------------------------------------------------------+
+   | Stage | {7, 2, 1, 1, 1, 10, 24, 8, 1, 1, 3, 23, 92, 23, 3, 1, 1,  |
+   | 1     | 8, 24, 10, 1, 1, 1, 2, 7}/256                             |
+   |       |                                                           |
+   | Stage | {85, 86, 85}/256                                          |
+   | 2     |                                                           |
+   |       |                                                           |
+   | Stage | {51, 51, 52, 51, 51}/256                                  |
+   | 3     |                                                           |
+   +-------+-----------------------------------------------------------+
+
+                        Table 6: Stereo Weight PDFs
+
+
+
+Valin, et al.                Standards Track                   [Page 40]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Then, use n, i0, and i2 to form two table indices, wi0 and wi1,
+   according to
+
+                             wi0 = i0 + 3*(n/5)
+                             wi1 = i2 + 3*(n%5)
+
+   where the division is integer division.  The range of these indices
+   is 0 to 14, inclusive.  Let w_Q13[i] be the i'th weight from Table 7.
+   Then, the two prediction weights, w0_Q13 and w1_Q13, are
+
+      w1_Q13 = w_Q13[wi1]
+               + (((w_Q13[wi1+1] - w_Q13[wi1])*6554) >> 16)*(2*i3 + 1)
+
+      w0_Q13 = w_Q13[wi0]
+               + (((w_Q13[wi0+1] - w_Q13[wi0])*6554) >> 16)*(2*i1 + 1)
+               - w1_Q13
+
+   N.B., w1_Q13 is computed first here, because w0_Q13 depends on it.
+   The constant 6554 is approximately 0.1 in Q16.  Although wi0 and wi1
+   only have 15 possible values, Table 7 contains 16 entries to allow
+   interpolation between entry wi0 and (wi0 + 1) (and likewise for wi1).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 41]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                         +-------+--------------+
+                         | Index | Weight (Q13) |
+                         +-------+--------------+
+                         | 0     |       -13732 |
+                         |       |              |
+                         | 1     |       -10050 |
+                         |       |              |
+                         | 2     |        -8266 |
+                         |       |              |
+                         | 3     |        -7526 |
+                         |       |              |
+                         | 4     |        -6500 |
+                         |       |              |
+                         | 5     |        -5000 |
+                         |       |              |
+                         | 6     |        -2950 |
+                         |       |              |
+                         | 7     |         -820 |
+                         |       |              |
+                         | 8     |          820 |
+                         |       |              |
+                         | 9     |         2950 |
+                         |       |              |
+                         | 10    |         5000 |
+                         |       |              |
+                         | 11    |         6500 |
+                         |       |              |
+                         | 12    |         7526 |
+                         |       |              |
+                         | 13    |         8266 |
+                         |       |              |
+                         | 14    |        10050 |
+                         |       |              |
+                         | 15    |        13732 |
+                         +-------+--------------+
+
+                       Table 7: Stereo Weight Table
+
+4.2.7.2.  Mid-Only Flag
+
+   A flag appears after the stereo prediction weights that indicates if
+   only the mid channel is coded for this time interval.  It appears
+   only when
+
+   o  This is a stereo Opus frame (see Section 3.1),
+
+   o  The current SILK frame corresponds to the mid channel, and
+
+
+
+
+Valin, et al.                Standards Track                   [Page 42]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  Either
+
+      *  This is a regular SILK frame where the VAD flags (see
+         Section 4.2.3) indicate that the corresponding side channel is
+         not active.
+
+      *  This is an LBRR frame where the LBRR flags (see Sections 4.2.3
+         and 4.2.4) indicate that the corresponding side channel is not
+         coded.
+
+   It is omitted when there are no stereo weights, for all of the same
+   reasons.  It is also omitted for a regular SILK frame when the VAD
+   flag of the corresponding side channel frame is set (indicating it is
+   active).  The side channel must be coded in this case, making the
+   mid-only flag redundant.  It is also omitted for an LBRR frame when
+   the corresponding LBRR flags indicate the side channel is coded.
+
+   When the flag is present, the decoder reads a single value using the
+   PDF in Table 8, as implemented in silk_stereo_decode_mid_only()
+   (stereo_decode_pred.c).  If the flag is set, then there is no
+   corresponding SILK frame for the side channel, the entire decoding
+   process for the side channel is skipped, and zeros are fed to the
+   stereo unmixing process (see Section 4.2.8) instead.  As stated
+   above, LBRR frames still include this flag when the LBRR flag
+   indicates that the side channel is not coded.  In that case, if this
+   flag is zero (indicating that there should be a side channel), then
+   Packet Loss Concealment (PLC, see Section 4.4) SHOULD be invoked to
+   recover a side channel signal.  Otherwise, the stereo image will
+   collapse.
+
+                             +---------------+
+                             | PDF           |
+                             +---------------+
+                             | {192, 64}/256 |
+                             +---------------+
+
+                        Table 8: Mid-only Flag PDF
+
+4.2.7.3.  Frame Type
+
+   Each SILK frame contains a single "frame type" symbol that jointly
+   codes the signal type and quantization offset type of the
+   corresponding frame.  If the current frame is a regular SILK frame
+   whose VAD bit was not set (an "inactive" frame), then the frame type
+   symbol takes on a value of either 0 or 1 and is decoded using the
+   first PDF in Table 9.  If the frame is an LBRR frame or a regular
+   SILK frame whose VAD flag was set (an "active" frame), then the value
+   of the symbol may range from 2 to 5, inclusive, and is decoded using
+
+
+
+Valin, et al.                Standards Track                   [Page 43]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   the second PDF in Table 9.  Table 10 translates between the value of
+   the frame type symbol and the corresponding signal type and
+   quantization offset type.
+
+                +----------+-----------------------------+
+                | VAD Flag | PDF                         |
+                +----------+-----------------------------+
+                | Inactive | {26, 230, 0, 0, 0, 0}/256   |
+                |          |                             |
+                | Active   | {0, 0, 24, 74, 148, 10}/256 |
+                +----------+-----------------------------+
+
+                         Table 9: Frame Type PDFs
+
+          +------------+-------------+--------------------------+
+          | Frame Type | Signal Type | Quantization Offset Type |
+          +------------+-------------+--------------------------+
+          | 0          | Inactive    |                      Low |
+          |            |             |                          |
+          | 1          | Inactive    |                     High |
+          |            |             |                          |
+          | 2          | Unvoiced    |                      Low |
+          |            |             |                          |
+          | 3          | Unvoiced    |                     High |
+          |            |             |                          |
+          | 4          | Voiced      |                      Low |
+          |            |             |                          |
+          | 5          | Voiced      |                     High |
+          +------------+-------------+--------------------------+
+
+    Table 10: Signal Type and Quantization Offset Type from Frame Type
+
+4.2.7.4.  Subframe Gains
+
+   A separate quantization gain is coded for each 5 ms subframe.  These
+   gains control the step size between quantization levels of the
+   excitation signal and, therefore, the quality of the reconstruction.
+   They are independent of and unrelated to the pitch contours coded for
+   voiced frames.  The quantization gains are themselves uniformly
+   quantized to 6 bits on a log scale, giving them a resolution of
+   approximately 1.369 dB and a range of approximately 1.94 dB to
+   88.21 dB.
+
+   The subframe gains are either coded independently, or relative to the
+   gain from the most recent coded subframe in the same channel.
+   Independent coding is used if and only if
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 44]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  This is the first subframe in the current SILK frame, and
+
+   o  Either
+
+      *  This is the first SILK frame of its type (LBRR or regular) for
+         this channel in the current Opus frame, or
+
+      *  The previous SILK frame of the same type (LBRR or regular) for
+         this channel in the same Opus frame was not coded.
+
+   In an independently coded subframe gain, the 3 most significant bits
+   of the quantization gain are decoded using a PDF selected from
+   Table 11 based on the decoded signal type (see Section 4.2.7.3).
+
+           +-------------+------------------------------------+
+           | Signal Type | PDF                                |
+           +-------------+------------------------------------+
+           | Inactive    | {32, 112, 68, 29, 12, 1, 1, 1}/256 |
+           |             |                                    |
+           | Unvoiced    | {2, 17, 45, 60, 62, 47, 19, 4}/256 |
+           |             |                                    |
+           | Voiced      | {1, 3, 26, 71, 94, 50, 9, 2}/256   |
+           +-------------+------------------------------------+
+
+        Table 11: PDFs for Independent Quantization Gain MSB Coding
+
+   The 3 least significant bits are decoded using a uniform PDF:
+
+                 +--------------------------------------+
+                 | PDF                                  |
+                 +--------------------------------------+
+                 | {32, 32, 32, 32, 32, 32, 32, 32}/256 |
+                 +--------------------------------------+
+
+        Table 12: PDF for Independent Quantization Gain LSB Coding
+
+   These 6 bits are combined to form a value, gain_index, between 0 and
+   63.  When the gain for the previous subframe is available, then the
+   current gain is limited as follows:
+
+             log_gain = max(gain_index, previous_log_gain - 16)
+
+   This may help some implementations limit the change in precision of
+   their internal LTP history.  The indices to which this clamp applies
+   cannot simply be removed from the codebook, because previous_log_gain
+   will not be available after packet loss.  The clamping is skipped
+   after a decoder reset, and in the side channel if the previous frame
+
+
+
+
+Valin, et al.                Standards Track                   [Page 45]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   in the side channel was not coded, since there is no value for
+   previous_log_gain available.  It MAY also be skipped after packet
+   loss.
+
+   For subframes that do not have an independent gain (including the
+   first subframe of frames not listed as using independent coding
+   above), the quantization gain is coded relative to the gain from the
+   previous subframe (in the same channel).  The PDF in Table 13 yields
+   a delta_gain_index value between 0 and 40, inclusive.
+
+   +-------------------------------------------------------------------+
+   | PDF                                                               |
+   +-------------------------------------------------------------------+
+   | {6, 5, 11, 31, 132, 21, 8, 4, 3, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, |
+   | 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,       |
+   | 1}/256                                                            |
+   +-------------------------------------------------------------------+
+
+             Table 13: PDF for Delta Quantization Gain Coding
+
+   The following formula translates this index into a quantization gain
+   for the current subframe using the gain from the previous subframe:
+
+     log_gain = clamp(0, max(2*delta_gain_index - 16,
+                        previous_log_gain + delta_gain_index - 4), 63)
+
+   silk_gains_dequant() (gain_quant.c) dequantizes log_gain for the k'th
+   subframe and converts it into a linear Q16 scale factor via
+
+         gain_Q16[k] = silk_log2lin((0x1D1C71*log_gain>>16) + 2090)
+
+   The function silk_log2lin() (log2lin.c) computes an approximation of
+   2**(inLog_Q7/128.0), where inLog_Q7 is its Q7 input.  Let i =
+   inLog_Q7>>7 be the integer part of inLogQ7 and f = inLog_Q7&127 be
+   the fractional part.  Then,
+
+               (1<<i) + ((-174*f*(128-f)>>16)+f)*((1<<i)>>7)
+
+   yields the approximate exponential.  The final Q16 gain values lies
+   between 81920 and 1686110208, inclusive (representing scale factors
+   of 1.25 to 25728, respectively).
+
+4.2.7.5.  Normalized Line Spectral Frequency (LSF) and Linear Predictive
+          Coding (LPC) Coefficients
+
+   A set of normalized Line Spectral Frequency (LSF) coefficients follow
+   the quantization gains in the bitstream and represent the Linear
+   Predictive Coding (LPC) coefficients for the current SILK frame.
+
+
+
+Valin, et al.                Standards Track                   [Page 46]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Once decoded, the normalized LSFs form an increasing list of Q15
+   values between 0 and 1.  These represent the interleaved zeros on the
+   upper half of the unit circle (between 0 and pi, hence "normalized")
+   in the standard decomposition [SPECTRAL-PAIRS] of the LPC filter into
+   a symmetric part and an anti-symmetric part (P and Q in
+   Section 4.2.7.5.6).  Because of non-linear effects in the decoding
+   process, an implementation SHOULD match the fixed-point arithmetic
+   described in this section exactly.  An encoder SHOULD also use the
+   same process.
+
+   The normalized LSFs are coded using a two-stage vector quantizer (VQ)
+   (Sections 4.2.7.5.1 and 4.2.7.5.2).  NB and MB frames use an order-10
+   predictor, while WB frames use an order-16 predictor.  Thus, each of
+   these two cases uses a different set of tables.  After reconstructing
+   the normalized LSFs (Section 4.2.7.5.3), the decoder runs them
+   through a stabilization process (Section 4.2.7.5.4), interpolates
+   them between frames (Section 4.2.7.5.5), converts them back into LPC
+   coefficients (Section 4.2.7.5.6), and then runs them through further
+   processes to limit the range of the coefficients (Section 4.2.7.5.7)
+   and the gain of the filter (Section 4.2.7.5.8).  All of this is
+   necessary to ensure the reconstruction process is stable.
+
+4.2.7.5.1.  Normalized LSF Stage 1 Decoding
+
+   The first VQ stage uses a 32-element codebook, coded with one of the
+   PDFs in Table 14, depending on the audio bandwidth and the signal
+   type of the current SILK frame.  This yields a single index, I1, for
+   the entire frame, which
+
+   1.  Indexes an element in a coarse codebook,
+
+   2.  Selects the PDFs for the second stage of the VQ, and
+
+   3.  Selects the prediction weights used to remove intra-frame
+       redundancy from the second stage.
+
+   The actual codebook elements are listed in Tables 23 and 24, but they
+   are not needed until the last stages of reconstructing the LSF
+   coefficients.
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 47]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +-----------+----------+--------------------------------------------+
+   | Audio     | Signal   | PDF                                        |
+   | Bandwidth | Type     |                                            |
+   +-----------+----------+--------------------------------------------+
+   | NB or MB  | Inactive | {44, 34, 30, 19, 21, 12, 11, 3, 3, 2, 16,  |
+   |           | or       | 2, 2, 1, 5, 2, 1, 3, 3, 1, 1, 2, 2, 2, 3,  |
+   |           | unvoiced | 1, 9, 9, 2, 7, 2, 1}/256                   |
+   |           |          |                                            |
+   | NB or MB  | Voiced   | {1, 10, 1, 8, 3, 8, 8, 14, 13, 14, 1, 14,  |
+   |           |          | 12, 13, 11, 11, 12, 11, 10, 10, 11, 8, 9,  |
+   |           |          | 8, 7, 8, 1, 1, 6, 1, 6, 5}/256             |
+   |           |          |                                            |
+   | WB        | Inactive | {31, 21, 3, 17, 1, 8, 17, 4, 1, 18, 16, 4, |
+   |           | or       | 2, 3, 1, 10, 1, 3, 16, 11, 16, 2, 2, 3, 2, |
+   |           | unvoiced | 11, 1, 4, 9, 8, 7, 3}/256                  |
+   |           |          |                                            |
+   | WB        | Voiced   | {1, 4, 16, 5, 18, 11, 5, 14, 15, 1, 3, 12, |
+   |           |          | 13, 14, 14, 6, 14, 12, 2, 6, 1, 12, 12,    |
+   |           |          | 11, 10, 3, 10, 5, 1, 1, 1, 3}/256          |
+   +-----------+----------+--------------------------------------------+
+
+         Table 14: PDFs for Normalized LSF Stage-1 Index Decoding
+
+4.2.7.5.2.  Normalized LSF Stage 2 Decoding
+
+   A total of 16 PDFs are available for the LSF residual in the second
+   stage: the 8 (a...h) for NB and MB frames given in Table 15, and the
+   8 (i...p) for WB frames given in Table 16.  Which PDF is used for
+   which coefficient is driven by the index, I1, decoded in the first
+   stage.  Table 17 lists the letter of the corresponding PDF for each
+   normalized LSF coefficient for NB and MB, and Table 18 lists the same
+   information for WB.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 48]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+            +----------+--------------------------------------+
+            | Codebook | PDF                                  |
+            +----------+--------------------------------------+
+            | a        | {1, 1, 1, 15, 224, 11, 1, 1, 1}/256  |
+            |          |                                      |
+            | b        | {1, 1, 2, 34, 183, 32, 1, 1, 1}/256  |
+            |          |                                      |
+            | c        | {1, 1, 4, 42, 149, 55, 2, 1, 1}/256  |
+            |          |                                      |
+            | d        | {1, 1, 8, 52, 123, 61, 8, 1, 1}/256  |
+            |          |                                      |
+            | e        | {1, 3, 16, 53, 101, 74, 6, 1, 1}/256 |
+            |          |                                      |
+            | f        | {1, 3, 17, 55, 90, 73, 15, 1, 1}/256 |
+            |          |                                      |
+            | g        | {1, 7, 24, 53, 74, 67, 26, 3, 1}/256 |
+            |          |                                      |
+            | h        | {1, 1, 18, 63, 78, 58, 30, 6, 1}/256 |
+            +----------+--------------------------------------+
+
+      Table 15: PDFs for NB/MB Normalized LSF Stage-2 Index Decoding
+
+           +----------+---------------------------------------+
+           | Codebook | PDF                                   |
+           +----------+---------------------------------------+
+           | i        | {1, 1, 1, 9, 232, 9, 1, 1, 1}/256     |
+           |          |                                       |
+           | j        | {1, 1, 2, 28, 186, 35, 1, 1, 1}/256   |
+           |          |                                       |
+           | k        | {1, 1, 3, 42, 152, 53, 2, 1, 1}/256   |
+           |          |                                       |
+           | l        | {1, 1, 10, 49, 126, 65, 2, 1, 1}/256  |
+           |          |                                       |
+           | m        | {1, 4, 19, 48, 100, 77, 5, 1, 1}/256  |
+           |          |                                       |
+           | n        | {1, 1, 14, 54, 100, 72, 12, 1, 1}/256 |
+           |          |                                       |
+           | o        | {1, 1, 15, 61, 87, 61, 25, 4, 1}/256  |
+           |          |                                       |
+           | p        | {1, 7, 21, 50, 77, 81, 17, 1, 1}/256  |
+           +----------+---------------------------------------+
+
+        Table 16: PDFs for WB Normalized LSF Stage-2 Index Decoding
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 49]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       +----+---------------------+
+                       | I1 | Coefficient         |
+                       +----+---------------------+
+                       |    | 0 1 2 3 4 5 6 7 8 9 |
+                       | 0  | a a a a a a a a a a |
+                       |    |                     |
+                       | 1  | b d b c c b c b b b |
+                       |    |                     |
+                       | 2  | c b b b b b b b b b |
+                       |    |                     |
+                       | 3  | b c c c c b c b b b |
+                       |    |                     |
+                       | 4  | c d d d d c c c c c |
+                       |    |                     |
+                       | 5  | a f d d c c c c b b |
+                       |    |                     |
+                       | g  | a c c c c c c c c b |
+                       |    |                     |
+                       | 7  | c d g e e e f e f f |
+                       |    |                     |
+                       | 8  | c e f f e f e g e e |
+                       |    |                     |
+                       | 9  | c e e h e f e f f e |
+                       |    |                     |
+                       | 10 | e d d d c d c c c c |
+                       |    |                     |
+                       | 11 | b f f g e f e f f f |
+                       |    |                     |
+                       | 12 | c h e g f f f f f f |
+                       |    |                     |
+                       | 13 | c h f f f f f g f e |
+                       |    |                     |
+                       | 14 | d d f e e f e f e e |
+                       |    |                     |
+                       | 15 | c d d f f e e e e e |
+                       |    |                     |
+                       | 16 | c e e g e f e f f f |
+                       |    |                     |
+                       | 17 | c f e g f f f e f e |
+                       |    |                     |
+                       | 18 | c h e f e f e f f f |
+                       |    |                     |
+                       | 19 | c f e g h g f g f e |
+                       |    |                     |
+                       | 20 | d g h e g f f g e f |
+                       |    |                     |
+                       | 21 | c h g e e e f e f f |
+                       |    |                     |
+
+
+
+Valin, et al.                Standards Track                   [Page 50]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       | 22 | e f f e g g f g f e |
+                       |    |                     |
+                       | 23 | c f f g f g e g e e |
+                       |    |                     |
+                       | 24 | e f f f d h e f f e |
+                       |    |                     |
+                       | 25 | c d e f f g e f f e |
+                       |    |                     |
+                       | 26 | c d c d d e c d d d |
+                       |    |                     |
+                       | 27 | b b c c c c c d c c |
+                       |    |                     |
+                       | 28 | e f f g g g f g e f |
+                       |    |                     |
+                       | 29 | d f f e e e e d d c |
+                       |    |                     |
+                       | 30 | c f d h f f e e f e |
+                       |    |                     |
+                       | 31 | e e f e f g f g f e |
+                       +----+---------------------+
+
+    Table 17: Codebook Selection for NB/MB Normalized LSF Stage-2 Index
+                                 Decoding
+
+          +----+------------------------------------------------+
+          | I1 | Coefficient                                    |
+          +----+------------------------------------------------+
+          |    | 0  1  2  3  4  5  6  7  8  9 10 11 12 13 14 15 |
+          |    |                                                |
+          | 0  | i  i  i  i  i  i  i  i  i  i  i  i  i  i  i  i |
+          |    |                                                |
+          | 1  | k  l  l  l  l  l  k  k  k  k  k  j  j  j  i  l |
+          |    |                                                |
+          | 2  | k  n  n  l  p  m  m  n  k  n  m  n  n  m  l  l |
+          |    |                                                |
+          | 3  | i  k  j  k  k  j  j  j  j  j  i  i  i  i  i  j |
+          |    |                                                |
+          | 4  | i  o  n  m  o  m  p  n  m  m  m  n  n  m  m  l |
+          |    |                                                |
+          | 5  | i  l  n  n  m  l  l  n  l  l  l  l  l  l  k  m |
+          |    |                                                |
+          | 6  | i  i  i  i  i  i  i  i  i  i  i  i  i  i  i  i |
+          |    |                                                |
+          | 7  | i  k  o  l  p  k  n  l  m  n  n  m  l  l  k  l |
+          |    |                                                |
+          | 8  | i  o  k  o  o  m  n  m  o  n  m  m  n  l  l  l |
+          |    |                                                |
+          | 9  | k  j  i  i  i  i  i  i  i  i  i  i  i  i  i  i |
+
+
+
+Valin, et al.                Standards Track                   [Page 51]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+          |    |                                                |
+          | 10 | i  j  i  i  i  i  i  i  i  i  i  i  i  i  i  j |
+          |    |                                                |
+          | 11 | k  k  l  m  n  l  l  l  l  l  l  l  k  k  j  l |
+          |    |                                                |
+          | 12 | k  k  l  l  m  l  l  l  l  l  l  l  l  k  j  l |
+          |    |                                                |
+          | 13 | l  m  m  m  o  m  m  n  l  n  m  m  n  m  l  m |
+          |    |                                                |
+          | 14 | i  o  m  n  m  p  n  k  o  n  p  m  m  l  n  l |
+          |    |                                                |
+          | 15 | i  j  i  j  j  j  j  j  j  j  i  i  i  i  j  i |
+          |    |                                                |
+          | 16 | j  o  n  p  n  m  n  l  m  n  m  m  m  l  l  m |
+          |    |                                                |
+          | 17 | j  l  l  m  m  l  l  n  k  l  l  n  n  n  l  m |
+          |    |                                                |
+          | 18 | k  l  l  k  k  k  l  k  j  k  j  k  j  j  j  m |
+          |    |                                                |
+          | 19 | i  k  l  n  l  l  k  k  k  j  j  i  i  i  i  i |
+          |    |                                                |
+          | 20 | l  m  l  n  l  l  k  k  j  j  j  j  j  k  k  m |
+          |    |                                                |
+          | 21 | k  o  l  p  p  m  n  m  n  l  n  l  l  k  l  l |
+          |    |                                                |
+          | 22 | k  l  n  o  o  l  n  l  m  m  l  l  l  l  k  m |
+          |    |                                                |
+          | 23 | j  l  l  m  m  m  m  l  n  n  n  l  j  j  j  j |
+          |    |                                                |
+          | 24 | k  n  l  o  o  m  p  m  m  n  l  m  m  l  l  l |
+          |    |                                                |
+          | 25 | i  o  j  j  i  i  i  i  i  i  i  i  i  i  i  i |
+          |    |                                                |
+          | 26 | i  o  o  l  n  k  n  n  l  m  m  p  p  m  m  m |
+          |    |                                                |
+          | 27 | l  l  p  l  n  m  l  l  l  k  k  l  l  l  k  l |
+          |    |                                                |
+          | 28 | i  i  j  i  i  i  k  j  k  j  j  k  k  k  j  j |
+          |    |                                                |
+          | 29 | i  l  k  n  l  l  k  l  k  j  i  i  j  i  i  j |
+          |    |                                                |
+          | 30 | l  n  n  m  p  n  l  l  k  l  k  k  j  i  j  i |
+          |    |                                                |
+          | 31 | k  l  n  l  m  l  l  l  k  j  k  o  m  i  i  i |
+          +----+------------------------------------------------+
+
+     Table 18: Codebook Selection for WB Normalized LSF Stage-2 Index
+                                 Decoding
+
+
+
+Valin, et al.                Standards Track                   [Page 52]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Decoding the second stage residual proceeds as follows.  For each
+   coefficient, the decoder reads a symbol using the PDF corresponding
+   to I1 from either Table 17 or Table 18, and subtracts 4 from the
+   result to give an index in the range -4 to 4, inclusive.  If the
+   index is either -4 or 4, it reads a second symbol using the PDF in
+   Table 19, and adds the value of this second symbol to the index,
+   using the same sign.  This gives the index, I2[k], a total range of
+   -10 to 10, inclusive.
+
+                     +-------------------------------+
+                     | PDF                           |
+                     +-------------------------------+
+                     | {156, 60, 24, 9, 4, 2, 1}/256 |
+                     +-------------------------------+
+
+         Table 19: PDF for Normalized LSF Index Extension Decoding
+
+   The decoded indices from both stages are translated back into
+   normalized LSF coefficients in silk_NLSF_decode() (NLSF_decode.c).
+   The stage-2 indices represent residuals after both the first stage of
+   the VQ and a separate backwards-prediction step.  The backwards
+   prediction process in the encoder subtracts a prediction from each
+   residual formed by a multiple of the coefficient that follows it.
+   The decoder must undo this process.  Table 20 contains lists of
+   prediction weights for each coefficient.  There are two lists for NB
+   and MB, and another two lists for WB, giving two possible prediction
+   weights for each coefficient.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 53]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                  +-------------+-----+-----+-----+-----+
+                  | Coefficient |   A |   B |   C |   D |
+                  +-------------+-----+-----+-----+-----+
+                  | 0           | 179 | 116 | 175 |  68 |
+                  |             |     |     |     |     |
+                  | 1           | 138 |  67 | 148 |  62 |
+                  |             |     |     |     |     |
+                  | 2           | 140 |  82 | 160 |  66 |
+                  |             |     |     |     |     |
+                  | 3           | 148 |  59 | 176 |  60 |
+                  |             |     |     |     |     |
+                  | 4           | 151 |  92 | 178 |  72 |
+                  |             |     |     |     |     |
+                  | 5           | 149 |  72 | 173 | 117 |
+                  |             |     |     |     |     |
+                  | 6           | 153 | 100 | 174 |  85 |
+                  |             |     |     |     |     |
+                  | 7           | 151 |  89 | 164 |  90 |
+                  |             |     |     |     |     |
+                  | 8           | 163 |  92 | 177 | 118 |
+                  |             |     |     |     |     |
+                  | 9           |     |     | 174 | 136 |
+                  |             |     |     |     |     |
+                  | 10          |     |     | 196 | 151 |
+                  |             |     |     |     |     |
+                  | 11          |     |     | 182 | 142 |
+                  |             |     |     |     |     |
+                  | 12          |     |     | 198 | 160 |
+                  |             |     |     |     |     |
+                  | 13          |     |     | 192 | 142 |
+                  |             |     |     |     |     |
+                  | 14          |     |     | 182 | 155 |
+                  +-------------+-----+-----+-----+-----+
+
+         Table 20: Prediction Weights for Normalized LSF Decoding
+
+   The prediction is undone using the procedure implemented in
+   silk_NLSF_residual_dequant() (NLSF_decode.c), which is as follows.
+   Each coefficient selects its prediction weight from one of the two
+   lists based on the stage-1 index, I1.  Table 21 gives the selections
+   for each coefficient for NB and MB, and Table 22 gives the selections
+   for WB.  Let d_LPC be the order of the codebook, i.e., 10 for NB and
+   MB, and 16 for WB, and let pred_Q8[k] be the weight for the k'th
+   coefficient selected by this process for 0 <= k < d_LPC-1.  Then, the
+   stage-2 residual for each coefficient is computed via
+
+       res_Q10[k] = (k+1 < d_LPC ? (res_Q10[k+1]*pred_Q8[k])>>8 : 0)
+                    + ((((I2[k]<<10) - sign(I2[k])*102)*qstep)>>16) ,
+
+
+
+Valin, et al.                Standards Track                   [Page 54]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   where qstep is the Q16 quantization step size, which is 11796 for NB
+   and MB and 9830 for WB (representing step sizes of approximately 0.18
+   and 0.15, respectively).
+
+                        +----+-------------------+
+                        | I1 | Coefficient       |
+                        +----+-------------------+
+                        |    | 0 1 2 3 4 5 6 7 8 |
+                        |    |                   |
+                        | 0  | A B A A A A A A A |
+                        |    |                   |
+                        | 1  | B A A A A A A A A |
+                        |    |                   |
+                        | 2  | A A A A A A A A A |
+                        |    |                   |
+                        | 3  | B B B A A A A B A |
+                        |    |                   |
+                        | 4  | A B A A A A A A A |
+                        |    |                   |
+                        | 5  | A B A A A A A A A |
+                        |    |                   |
+                        | 6  | B A B B A A A B A |
+                        |    |                   |
+                        | 7  | A B B A A B B A A |
+                        |    |                   |
+                        | 8  | A A B B A B A B B |
+                        |    |                   |
+                        | 9  | A A B B A A B B B |
+                        |    |                   |
+                        | 10 | A A A A A A A A A |
+                        |    |                   |
+                        | 11 | A B A B B B B B A |
+                        |    |                   |
+                        | 12 | A B A B B B B B A |
+                        |    |                   |
+                        | 13 | A B B B B B B B A |
+                        |    |                   |
+                        | 14 | B A B B A B B B B |
+                        |    |                   |
+                        | 15 | A B B B B B A B A |
+                        |    |                   |
+                        | 16 | A A B B A B A B A |
+                        |    |                   |
+                        | 17 | A A B B B A B B B |
+                        |    |                   |
+                        | 18 | A B B A A B B B A |
+                        |    |                   |
+                        | 19 | A A A B B B A B A |
+
+
+
+Valin, et al.                Standards Track                   [Page 55]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                        |    |                   |
+                        | 20 | A B B A A B A B A |
+                        |    |                   |
+                        | 21 | A B B A A A B B A |
+                        |    |                   |
+                        | 22 | A A A A A B B B B |
+                        |    |                   |
+                        | 23 | A A B B A A A B B |
+                        |    |                   |
+                        | 24 | A A A B A B B B B |
+                        |    |                   |
+                        | 25 | A B B B B B B B A |
+                        |    |                   |
+                        | 26 | A A A A A A A A A |
+                        |    |                   |
+                        | 27 | A A A A A A A A A |
+                        |    |                   |
+                        | 28 | A A B A B B A B A |
+                        |    |                   |
+                        | 29 | B A A B A A A A A |
+                        |    |                   |
+                        | 30 | A A A B B A B A B |
+                        |    |                   |
+                        | 31 | B A B B A B B B B |
+                        +----+-------------------+
+
+      Table 21: Prediction Weight Selection for NB/MB Normalized LSF
+                                 Decoding
+
+           +----+---------------------------------------------+
+           | I1 | Coefficient                                 |
+           +----+---------------------------------------------+
+           |    | 0  1  2  3  4  5  6  7  8  9 10 11 12 13 14 |
+           |    |                                             |
+           | 0  | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+           |    |                                             |
+           | 1  | C  C  C  C  C  C  C  C  C  C  C  C  C  C  C |
+           |    |                                             |
+           | 2  | C  C  D  C  C  D  D  D  C  D  D  D  D  C  C |
+           |    |                                             |
+           | 3  | C  C  C  C  C  C  C  C  C  C  C  C  D  C  C |
+           |    |                                             |
+           | 4  | C  D  D  C  D  C  D  D  C  D  D  D  D  D  C |
+           |    |                                             |
+           | 5  | C  C  D  C  C  C  C  C  C  C  C  C  C  C  C |
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 56]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+           |    |                                             |
+           | 6  | D  C  C  C  C  C  C  C  C  C  C  D  C  D  C |
+           |    |                                             |
+           | 7  | C  D  D  C  C  C  D  C  D  D  D  C  D  C  D |
+           |    |                                             |
+           | 8  | C  D  C  D  D  C  D  C  D  C  D  D  D  D  D |
+           |    |                                             |
+           | 9  | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+           |    |                                             |
+           | 10 | C  D  C  C  C  C  C  C  C  C  C  C  C  C  C |
+           |    |                                             |
+           | 11 | C  C  D  C  D  D  D  D  D  D  D  C  D  C  C |
+           |    |                                             |
+           | 12 | C  C  D  C  C  D  C  D  C  D  C  C  D  C  C |
+           |    |                                             |
+           | 13 | C  C  C  C  D  D  C  D  C  D  D  D  D  C  C |
+           |    |                                             |
+           | 14 | C  D  C  C  C  D  D  C  D  D  D  C  D  D  D |
+           |    |                                             |
+           | 15 | C  C  D  D  C  C  C  C  C  C  C  C  D  D  C |
+           |    |                                             |
+           | 16 | C  D  D  C  D  C  D  D  D  D  D  C  D  C  C |
+           |    |                                             |
+           | 17 | C  C  D  C  C  C  C  D  C  C  D  D  D  C  C |
+           |    |                                             |
+           | 18 | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+           |    |                                             |
+           | 19 | C  C  C  C  C  C  C  C  C  C  C  C  D  C  C |
+           |    |                                             |
+           | 20 | C  C  C  C  C  C  C  C  C  C  C  C  C  C  C |
+           |    |                                             |
+           | 21 | C  D  C  D  C  D  D  C  D  C  D  C  D  D  C |
+           |    |                                             |
+           | 22 | C  C  D  D  D  D  C  D  D  C  C  D  D  C  C |
+           |    |                                             |
+           | 23 | C  D  D  C  D  C  D  C  D  C  C  C  C  D  C |
+           |    |                                             |
+           | 24 | C  C  C  D  D  C  D  C  D  D  D  D  D  D  D |
+           |    |                                             |
+           | 25 | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+           |    |                                             |
+           | 26 | C  D  D  C  C  C  D  D  C  C  D  D  D  D  D |
+           |    |                                             |
+           | 27 | C  C  C  C  C  D  C  D  D  D  D  C  D  D  D |
+           |    |                                             |
+           | 28 | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+           |    |                                             |
+           | 29 | C  C  C  C  C  C  C  C  C  C  C  C  C  C  D |
+
+
+
+Valin, et al.                Standards Track                   [Page 57]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+           |    |                                             |
+           | 30 | D  C  C  C  C  C  C  C  C  C  C  D  C  C  C |
+           |    |                                             |
+           | 31 | C  C  D  C  C  D  D  D  C  C  D  C  C  D  C |
+           +----+---------------------------------------------+
+
+   Table 22: Prediction Weight Selection for WB Normalized LSF Decoding
+
+4.2.7.5.3.  Reconstructing the Normalized LSF Coefficients
+
+   Once the stage-1 index I1 and the stage-2 residual res_Q10[] have
+   been decoded, the final normalized LSF coefficients can be
+   reconstructed.
+
+   The spectral distortion introduced by the quantization of each LSF
+   coefficient varies, so the stage-2 residual is weighted accordingly,
+   using the low-complexity Inverse Harmonic Mean Weighting (IHMW)
+   function proposed in [LAROIA-ICASSP].  The weights are derived
+   directly from the stage-1 codebook vector.  Let cb1_Q8[k] be the k'th
+   entry of the stage-1 codebook vector from Table 23 or Table 24.
+   Then, for 0 <= k < d_LPC, the following expression computes the
+   square of the weight as a Q18 value:
+
+
+            w2_Q18[k] = (1024/(cb1_Q8[k] - cb1_Q8[k-1])
+                         + 1024/(cb1_Q8[k+1] - cb1_Q8[k])) << 16
+
+
+   where cb1_Q8[-1] = 0 and cb1_Q8[d_LPC] = 256, and the division is
+   integer division.  This is reduced to an unsquared, Q9 value using
+   the following square-root approximation:
+
+                 i = ilog(w2_Q18[k])
+                 f = (w2_Q18[k]>>(i-8)) & 127
+                 y = ((i&1) ? 32768 : 46214) >> ((32-i)>>1)
+                 w_Q9[k] = y + ((213*f*y)>>16)
+
+   The constant 46214 here is approximately the square root of 2 in Q15.
+   The cb1_Q8[] vector completely determines these weights, and they may
+   be tabulated and stored as 13-bit unsigned values (with a range of
+   1819 to 5227, inclusive) to avoid computing them when decoding.  The
+   reference implementation already requires code to compute these
+   weights on unquantized coefficients in the encoder, in
+   silk_NLSF_VQ_weights_laroia() (NLSF_VQ_weights_laroia.c) and its
+   callers, so it reuses that code in the decoder instead of using a
+   pre-computed table to reduce the amount of ROM required.
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 58]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+              +----+----------------------------------------+
+              | I1 | Codebook (Q8)                          |
+              +----+----------------------------------------+
+              |    |  0   1   2   3   4   5   6   7   8   9 |
+              |    |                                        |
+              | 0  | 12  35  60  83 108 132 157 180 206 228 |
+              |    |                                        |
+              | 1  | 15  32  55  77 101 125 151 175 201 225 |
+              |    |                                        |
+              | 2  | 19  42  66  89 114 137 162 184 209 230 |
+              |    |                                        |
+              | 3  | 12  25  50  72  97 120 147 172 200 223 |
+              |    |                                        |
+              | 4  | 26  44  69  90 114 135 159 180 205 225 |
+              |    |                                        |
+              | 5  | 13  22  53  80 106 130 156 180 205 228 |
+              |    |                                        |
+              | 6  | 15  25  44  64  90 115 142 168 196 222 |
+              |    |                                        |
+              | 7  | 19  24  62  82 100 120 145 168 190 214 |
+              |    |                                        |
+              | 8  | 22  31  50  79 103 120 151 170 203 227 |
+              |    |                                        |
+              | 9  | 21  29  45  65 106 124 150 171 196 224 |
+              |    |                                        |
+              | 10 | 30  49  75  97 121 142 165 186 209 229 |
+              |    |                                        |
+              | 11 | 19  25  52  70  93 116 143 166 192 219 |
+              |    |                                        |
+              | 12 | 26  34  62  75  97 118 145 167 194 217 |
+              |    |                                        |
+              | 13 | 25  33  56  70  91 113 143 165 196 223 |
+              |    |                                        |
+              | 14 | 21  34  51  72  97 117 145 171 196 222 |
+              |    |                                        |
+              | 15 | 20  29  50  67  90 117 144 168 197 221 |
+              |    |                                        |
+              | 16 | 22  31  48  66  95 117 146 168 196 222 |
+              |    |                                        |
+              | 17 | 24  33  51  77 116 134 158 180 200 224 |
+              |    |                                        |
+              | 18 | 21  28  70  87 106 124 149 170 194 217 |
+              |    |                                        |
+              | 19 | 26  33  53  64  83 117 152 173 204 225 |
+              |    |                                        |
+              | 20 | 27  34  65  95 108 129 155 174 210 225 |
+              |    |                                        |
+              | 21 | 20  26  72  99 113 131 154 176 200 219 |
+
+
+
+Valin, et al.                Standards Track                   [Page 59]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+              |    |                                        |
+              | 22 | 34  43  61  78  93 114 155 177 205 229 |
+              |    |                                        |
+              | 23 | 23  29  54  97 124 138 163 179 209 229 |
+              |    |                                        |
+              | 24 | 30  38  56  89 118 129 158 178 200 231 |
+              |    |                                        |
+              | 25 | 21  29  49  63  85 111 142 163 193 222 |
+              |    |                                        |
+              | 26 | 27  48  77 103 133 158 179 196 215 232 |
+              |    |                                        |
+              | 27 | 29  47  74  99 124 151 176 198 220 237 |
+              |    |                                        |
+              | 28 | 33  42  61  76  93 121 155 174 207 225 |
+              |    |                                        |
+              | 29 | 29  53  87 112 136 154 170 188 208 227 |
+              |    |                                        |
+              | 30 | 24  30  52  84 131 150 166 186 203 229 |
+              |    |                                        |
+              | 31 | 37  48  64  84 104 118 156 177 201 230 |
+              +----+----------------------------------------+
+
+          Table 23: NB/MB Normalized LSF Stage-1 Codebook Vectors
+
+    +----+------------------------------------------------------------+
+    | I1 | Codebook (Q8)                                              |
+    +----+------------------------------------------------------------+
+    |    |  0  1  2  3  4   5   6   7   8   9  10  11  12  13  14  15 |
+    |    |                                                            |
+    | 0  |  7 23 38 54 69  85 100 116 131 147 162 178 193 208 223 239 |
+    |    |                                                            |
+    | 1  | 13 25 41 55 69  83  98 112 127 142 157 171 187 203 220 236 |
+    |    |                                                            |
+    | 2  | 15 21 34 51 61  78  92 106 126 136 152 167 185 205 225 240 |
+    |    |                                                            |
+    | 3  | 10 21 36 50 63  79  95 110 126 141 157 173 189 205 221 237 |
+    |    |                                                            |
+    | 4  | 17 20 37 51 59  78  89 107 123 134 150 164 184 205 224 240 |
+    |    |                                                            |
+    | 5  | 10 15 32 51 67  81  96 112 129 142 158 173 189 204 220 236 |
+    |    |                                                            |
+    | 6  |  8 21 37 51 65  79  98 113 126 138 155 168 179 192 209 218 |
+    |    |                                                            |
+    | 7  | 12 15 34 55 63  78  87 108 118 131 148 167 185 203 219 236 |
+    |    |                                                            |
+    | 8  | 16 19 32 36 56  79  91 108 118 136 154 171 186 204 220 237 |
+    |    |                                                            |
+    | 9  | 11 28 43 58 74  89 105 120 135 150 165 180 196 211 226 241 |
+
+
+
+Valin, et al.                Standards Track                   [Page 60]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+    |    |                                                            |
+    | 10 |  6 16 33 46 60  75  92 107 123 137 156 169 185 199 214 225 |
+    |    |                                                            |
+    | 11 | 11 19 30 44 57  74  89 105 121 135 152 169 186 202 218 234 |
+    |    |                                                            |
+    | 12 | 12 19 29 46 57  71  88 100 120 132 148 165 182 199 216 233 |
+    |    |                                                            |
+    | 13 | 17 23 35 46 56  77  92 106 123 134 152 167 185 204 222 237 |
+    |    |                                                            |
+    | 14 | 14 17 45 53 63  75  89 107 115 132 151 171 188 206 221 240 |
+    |    |                                                            |
+    | 15 |  9 16 29 40 56  71  88 103 119 137 154 171 189 205 222 237 |
+    |    |                                                            |
+    | 16 | 16 19 36 48 57  76  87 105 118 132 150 167 185 202 218 236 |
+    |    |                                                            |
+    | 17 | 12 17 29 54 71  81  94 104 126 136 149 164 182 201 221 237 |
+    |    |                                                            |
+    | 18 | 15 28 47 62 79  97 115 129 142 155 168 180 194 208 223 238 |
+    |    |                                                            |
+    | 19 |  8 14 30 45 62  78  94 111 127 143 159 175 192 207 223 239 |
+    |    |                                                            |
+    | 20 | 17 30 49 62 79  92 107 119 132 145 160 174 190 204 220 235 |
+    |    |                                                            |
+    | 21 | 14 19 36 45 61  76  91 108 121 138 154 172 189 205 222 238 |
+    |    |                                                            |
+    | 22 | 12 18 31 45 60  76  91 107 123 138 154 171 187 204 221 236 |
+    |    |                                                            |
+    | 23 | 13 17 31 43 53  70  83 103 114 131 149 167 185 203 220 237 |
+    |    |                                                            |
+    | 24 | 17 22 35 42 58  78  93 110 125 139 155 170 188 206 224 240 |
+    |    |                                                            |
+    | 25 |  8 15 34 50 67  83  99 115 131 146 162 178 193 209 224 239 |
+    |    |                                                            |
+    | 26 | 13 16 41 66 73  86  95 111 128 137 150 163 183 206 225 241 |
+    |    |                                                            |
+    | 27 | 17 25 37 52 63  75  92 102 119 132 144 160 175 191 212 231 |
+    |    |                                                            |
+    | 28 | 19 31 49 65 83 100 117 133 147 161 174 187 200 213 227 242 |
+    |    |                                                            |
+    | 29 | 18 31 52 68 88 103 117 126 138 149 163 177 192 207 223 239 |
+    |    |                                                            |
+    | 30 | 16 29 47 61 76  90 106 119 133 147 161 176 193 209 224 240 |
+    |    |                                                            |
+    | 31 | 15 21 35 50 61  73  86  97 110 119 129 141 175 198 218 237 |
+    +----+------------------------------------------------------------+
+
+           Table 24: WB Normalized LSF Stage-1 Codebook Vectors
+
+
+
+
+Valin, et al.                Standards Track                   [Page 61]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Given the stage-1 codebook entry cb1_Q8[], the stage-2 residual
+   res_Q10[], and their corresponding weights, w_Q9[], the reconstructed
+   normalized LSF coefficients are
+
+      NLSF_Q15[k] = clamp(0,
+                     (cb1_Q8[k]<<7) + (res_Q10[k]<<14)/w_Q9[k], 32767)
+
+   where the division is integer division.  However, nothing in either
+   the reconstruction process or the quantization process in the encoder
+   thus far guarantees that the coefficients are monotonically
+   increasing and separated well enough to ensure a stable filter
+   [KABAL86].  When using the reference encoder, roughly 2% of frames
+   violate this constraint.  The next section describes a stabilization
+   procedure used to make these guarantees.
+
+4.2.7.5.4.  Normalized LSF Stabilization
+
+   The normalized LSF stabilization procedure is implemented in
+   silk_NLSF_stabilize() (NLSF_stabilize.c).  This process ensures that
+   consecutive values of the normalized LSF coefficients, NLSF_Q15[],
+   are spaced some minimum distance apart (predetermined to be the 0.01
+   percentile of a large training set).  Table 25 gives the minimum
+   spacings for NB and MB and those for WB, where row k is the minimum
+   allowed value of NLSF_Q15[k]-NLSF_Q15[k-1].  For the purposes of
+   computing this spacing for the first and last coefficient,
+   NLSF_Q15[-1] is taken to be 0 and NLSF_Q15[d_LPC] is taken to be
+   32768.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 62]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                     +-------------+-----------+-----+
+                     | Coefficient | NB and MB |  WB |
+                     +-------------+-----------+-----+
+                     | 0           |       250 | 100 |
+                     |             |           |     |
+                     | 1           |         3 |   3 |
+                     |             |           |     |
+                     | 2           |         6 |  40 |
+                     |             |           |     |
+                     | 3           |         3 |   3 |
+                     |             |           |     |
+                     | 4           |         3 |   3 |
+                     |             |           |     |
+                     | 5           |         3 |   3 |
+                     |             |           |     |
+                     | 6           |         4 |   5 |
+                     |             |           |     |
+                     | 7           |         3 |  14 |
+                     |             |           |     |
+                     | 8           |         3 |  14 |
+                     |             |           |     |
+                     | 9           |         3 |  10 |
+                     |             |           |     |
+                     | 10          |       461 |  11 |
+                     |             |           |     |
+                     | 11          |           |   3 |
+                     |             |           |     |
+                     | 12          |           |   8 |
+                     |             |           |     |
+                     | 13          |           |   9 |
+                     |             |           |     |
+                     | 14          |           |   7 |
+                     |             |           |     |
+                     | 15          |           |   3 |
+                     |             |           |     |
+                     | 16          |           | 347 |
+                     +-------------+-----------+-----+
+
+         Table 25: Minimum Spacing for Normalized LSF Coefficients
+
+   The procedure starts off by trying to make small adjustments that
+   attempt to minimize the amount of distortion introduced.  After 20
+   such adjustments, it falls back to a more direct method that
+   guarantees the constraints are enforced but may require large
+   adjustments.
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 63]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Let NDeltaMin_Q15[k] be the minimum required spacing for the current
+   audio bandwidth from Table 25.  First, the procedure finds the index
+   i where NLSF_Q15[i] - NLSF_Q15[i-1] - NDeltaMin_Q15[i] is the
+   smallest, breaking ties by using the lower value of i.  If this value
+   is non-negative, then the stabilization stops; the coefficients
+   satisfy all the constraints.  Otherwise, if i == 0, it sets
+   NLSF_Q15[0] to NDeltaMin_Q15[0], and if i == d_LPC, it sets
+   NLSF_Q15[d_LPC-1] to (32768 - NDeltaMin_Q15[d_LPC]).  For all other
+   values of i, both NLSF_Q15[i-1] and NLSF_Q15[i] are updated as
+   follows:
+
+                                             i-1
+                                             __
+    min_center_Q15 = (NDeltaMin_Q15[i]>>1) + \  NDeltaMin_Q15[k]
+                                             /_
+                                             k=0
+                                                    d_LPC
+                                                     __
+    max_center_Q15 = 32768 - (NDeltaMin_Q15[i]>>1) - \  NDeltaMin_Q15[k]
+                                                     /_
+                                                    k=i+1
+   center_freq_Q15 = clamp(min_center_Q15[i],
+                           (NLSF_Q15[i-1] + NLSF_Q15[i] + 1)>>1
+                           max_center_Q15[i])
+
+    NLSF_Q15[i-1] = center_freq_Q15 - (NDeltaMin_Q15[i]>>1)
+
+      NLSF_Q15[i] = NLSF_Q15[i-1] + NDeltaMin_Q15[i]
+
+   Then, the procedure repeats again, until it has either executed 20
+   times or stopped because the coefficients satisfy all the
+   constraints.
+
+   After the 20th repetition of the above procedure, the following
+   fallback procedure executes once.  First, the values of NLSF_Q15[k]
+   for 0 <= k < d_LPC are sorted in ascending order.  Then, for each
+   value of k from 0 to d_LPC-1, NLSF_Q15[k] is set to
+
+             max(NLSF_Q15[k], NLSF_Q15[k-1] + NDeltaMin_Q15[k])
+
+   Next, for each value of k from d_LPC-1 down to 0, NLSF_Q15[k] is set
+   to
+
+            min(NLSF_Q15[k], NLSF_Q15[k+1] - NDeltaMin_Q15[k+1])
+
+   There is no need to check if the coefficients satisfy all the
+   constraints before applying this fallback procedure.  If they do,
+   then it will not change their values.
+
+
+
+Valin, et al.                Standards Track                   [Page 64]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.2.7.5.5.  Normalized LSF Interpolation
+
+   For 20 ms SILK frames, the first half of the frame (i.e., the first
+   two subframes) may use normalized LSF coefficients that are
+   interpolated between the decoded LSFs for the most recent coded frame
+   (in the same channel) and the current frame.  A Q2 interpolation
+   factor follows the LSF coefficient indices in the bitstream, which is
+   decoded using the PDF in Table 26.  This happens in
+   silk_decode_indices() (decode_indices.c).  After either
+
+   o  An uncoded regular SILK frame in the side channel, or
+
+   o  A decoder reset (see Section 4.5.2),
+
+   the decoder still decodes this factor, but ignores its value and
+   always uses 4 instead.  For 10 ms SILK frames, this factor is not
+   stored at all.
+
+                       +---------------------------+
+                       | PDF                       |
+                       +---------------------------+
+                       | {13, 22, 29, 11, 181}/256 |
+                       +---------------------------+
+
+           Table 26: PDF for Normalized LSF Interpolation Index
+
+   Let n2_Q15[k] be the normalized LSF coefficients decoded by the
+   procedure in Section 4.2.7.5, n0_Q15[k] be the LSF coefficients
+   decoded for the prior frame, and w_Q2 be the interpolation factor.
+   Then, the normalized LSF coefficients used for the first half of a
+   20 ms frame, n1_Q15[k], are
+
+        n1_Q15[k] = n0_Q15[k] + (w_Q2*(n2_Q15[k] - n0_Q15[k]) >> 2)
+
+   This interpolation is performed in silk_decode_parameters()
+   (decode_parameters.c).
+
+4.2.7.5.6.  Converting Normalized LSFs to LPC Coefficients
+
+   Any LPC filter A(z) can be split into a symmetric part P(z) and an
+   anti-symmetric part Q(z) such that
+
+                          d_LPC
+                           __         -k   1
+                A(z) = 1 - \  a[k] * z   = - * (P(z) + Q(z))
+                           /_              2
+                           k=1
+
+
+
+
+Valin, et al.                Standards Track                   [Page 65]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   with
+
+                                     -d_LPC-1      -1
+                      P(z) = A(z) + z         * A(z  )
+
+                                     -d_LPC-1      -1
+                      Q(z) = A(z) - z         * A(z  )
+
+   The even normalized LSF coefficients correspond to a pair of
+   conjugate roots of P(z), while the odd coefficients correspond to a
+   pair of conjugate roots of Q(z), all of which lie on the unit circle.
+   In addition, P(z) has a root at pi and Q(z) has a root at 0.  Thus,
+   they may be reconstructed mathematically from a set of normalized LSF
+   coefficients, n[k], as
+
+                          d_LPC/2-1
+                      -1     ___                        -1    -2
+         P(z) = (1 + z  ) *  | |  (1 - 2*cos(pi*n[2*k])*z  + z  )
+                             k=0
+
+                          d_LPC/2-1
+                      -1     ___                          -1    -2
+         Q(z) = (1 - z  ) *  | |  (1 - 2*cos(pi*n[2*k+1])*z  + z  )
+                             k=0
+
+   However, SILK performs this reconstruction using a fixed-point
+   approximation so that all decoders can reproduce it in a bit-exact
+   manner to avoid prediction drift.  The function silk_NLSF2A()
+   (NLSF2A.c) implements this procedure.
+
+   To start, it approximates cos(pi*n[k]) using a table lookup with
+   linear interpolation.  The encoder SHOULD use the inverse of this
+   piecewise linear approximation, rather than the true inverse of the
+   cosine function, when deriving the normalized LSF coefficients.
+   These values are also re-ordered to improve numerical accuracy when
+   constructing the LPC polynomials.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 66]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                     +-------------+-----------+----+
+                     | Coefficient | NB and MB | WB |
+                     +-------------+-----------+----+
+                     | 0           |         0 |  0 |
+                     |             |           |    |
+                     | 1           |         9 | 15 |
+                     |             |           |    |
+                     | 2           |         6 |  8 |
+                     |             |           |    |
+                     | 3           |         3 |  7 |
+                     |             |           |    |
+                     | 4           |         4 |  4 |
+                     |             |           |    |
+                     | 5           |         5 | 11 |
+                     |             |           |    |
+                     | 6           |         8 | 12 |
+                     |             |           |    |
+                     | 7           |         1 |  3 |
+                     |             |           |    |
+                     | 8           |         2 |  2 |
+                     |             |           |    |
+                     | 9           |         7 | 13 |
+                     |             |           |    |
+                     | 10          |           | 10 |
+                     |             |           |    |
+                     | 11          |           |  5 |
+                     |             |           |    |
+                     | 12          |           |  6 |
+                     |             |           |    |
+                     | 13          |           |  9 |
+                     |             |           |    |
+                     | 14          |           | 14 |
+                     |             |           |    |
+                     | 15          |           |  1 |
+                     +-------------+-----------+----+
+
+             Table 27: LSF Ordering for Polynomial Evaluation
+
+   The top 7 bits of each normalized LSF coefficient index a value in
+   the table, and the next 8 bits interpolate between it and the next
+   value.  Let i = (n[k] >> 8) be the integer index and f = (n[k] & 255)
+   be the fractional part of a given coefficient.  Then, the re-ordered,
+   approximated cosine, c_Q17[ordering[k]], is
+
+       c_Q17[ordering[k]] = (cos_Q12[i]*256
+                             + (cos_Q12[i+1]-cos_Q12[i])*f + 4) >> 3
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 67]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   where ordering[k] is the k'th entry of the column of Table 27
+   corresponding to the current audio bandwidth and cos_Q12[i] is the
+   i'th entry of Table 28.
+
+                  +-----+-------+-------+-------+-------+
+                  |   i |    +0 |    +1 |    +2 |    +3 |
+                  +-----+-------+-------+-------+-------+
+                  |   0 |  4096 |  4095 |  4091 |  4085 |
+                  |     |       |       |       |       |
+                  |   4 |  4076 |  4065 |  4052 |  4036 |
+                  |     |       |       |       |       |
+                  |   8 |  4017 |  3997 |  3973 |  3948 |
+                  |     |       |       |       |       |
+                  |  12 |  3920 |  3889 |  3857 |  3822 |
+                  |     |       |       |       |       |
+                  |  16 |  3784 |  3745 |  3703 |  3659 |
+                  |     |       |       |       |       |
+                  |  20 |  3613 |  3564 |  3513 |  3461 |
+                  |     |       |       |       |       |
+                  |  24 |  3406 |  3349 |  3290 |  3229 |
+                  |     |       |       |       |       |
+                  |  28 |  3166 |  3102 |  3035 |  2967 |
+                  |     |       |       |       |       |
+                  |  32 |  2896 |  2824 |  2751 |  2676 |
+                  |     |       |       |       |       |
+                  |  36 |  2599 |  2520 |  2440 |  2359 |
+                  |     |       |       |       |       |
+                  |  40 |  2276 |  2191 |  2106 |  2019 |
+                  |     |       |       |       |       |
+                  |  44 |  1931 |  1842 |  1751 |  1660 |
+                  |     |       |       |       |       |
+                  |  48 |  1568 |  1474 |  1380 |  1285 |
+                  |     |       |       |       |       |
+                  |  52 |  1189 |  1093 |   995 |   897 |
+                  |     |       |       |       |       |
+                  |  56 |   799 |   700 |   601 |   501 |
+                  |     |       |       |       |       |
+                  |  60 |   401 |   301 |   201 |   101 |
+                  |     |       |       |       |       |
+                  |  64 |     0 |  -101 |  -201 |  -301 |
+                  |     |       |       |       |       |
+                  |  68 |  -401 |  -501 |  -601 |  -700 |
+                  |     |       |       |       |       |
+                  |  72 |  -799 |  -897 |  -995 | -1093 |
+                  |     |       |       |       |       |
+                  |  76 | -1189 | -1285 | -1380 | -1474 |
+                  |     |       |       |       |       |
+                  |  80 | -1568 | -1660 | -1751 | -1842 |
+
+
+
+Valin, et al.                Standards Track                   [Page 68]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                  |     |       |       |       |       |
+                  |  84 | -1931 | -2019 | -2106 | -2191 |
+                  |     |       |       |       |       |
+                  |  88 | -2276 | -2359 | -2440 | -2520 |
+                  |     |       |       |       |       |
+                  |  92 | -2599 | -2676 | -2751 | -2824 |
+                  |     |       |       |       |       |
+                  |  96 | -2896 | -2967 | -3035 | -3102 |
+                  |     |       |       |       |       |
+                  | 100 | -3166 | -3229 | -3290 | -3349 |
+                  |     |       |       |       |       |
+                  | 104 | -3406 | -3461 | -3513 | -3564 |
+                  |     |       |       |       |       |
+                  | 108 | -3613 | -3659 | -3703 | -3745 |
+                  |     |       |       |       |       |
+                  | 112 | -3784 | -3822 | -3857 | -3889 |
+                  |     |       |       |       |       |
+                  | 116 | -3920 | -3948 | -3973 | -3997 |
+                  |     |       |       |       |       |
+                  | 120 | -4017 | -4036 | -4052 | -4065 |
+                  |     |       |       |       |       |
+                  | 124 | -4076 | -4085 | -4091 | -4095 |
+                  |     |       |       |       |       |
+                  | 128 | -4096 |       |       |       |
+                  +-----+-------+-------+-------+-------+
+
+               Table 28: Q12 Cosine Table for LSF Conversion
+
+   Given the list of cosine values, silk_NLSF2A_find_poly() (NLSF2A.c)
+   computes the coefficients of P and Q, described here via a simple
+   recurrence.  Let p_Q16[k][j] and q_Q16[k][j] be the coefficients of
+   the products of the first (k+1) root pairs for P and Q, with j
+   indexing the coefficient number.  Only the first (k+2) coefficients
+   are needed, as the products are symmetric.  Let
+   p_Q16[0][0] = q_Q16[0][0] = 1<<16, p_Q16[0][1] = -c_Q17[0],
+   q_Q16[0][1] = -c_Q17[1], and d2 = d_LPC/2.  As boundary conditions,
+   assume p_Q16[k][j] = q_Q16[k][j] = 0 for all j < 0.  Also, assume
+   p_Q16[k][k+2] = p_Q16[k][k] and q_Q16[k][k+2] = q_Q16[k][k] (because
+   of the symmetry).  Then, for 0 < k < d2 and 0 <= j <= k+1,
+
+        p_Q16[k][j] = p_Q16[k-1][j] + p_Q16[k-1][j-2]
+                      - ((c_Q17[2*k]*p_Q16[k-1][j-1] + 32768)>>16)
+
+        q_Q16[k][j] = q_Q16[k-1][j] + q_Q16[k-1][j-2]
+                      - ((c_Q17[2*k+1]*q_Q16[k-1][j-1] + 32768)>>16)
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 69]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The use of Q17 values for the cosine terms in an otherwise Q16
+   expression implicitly scales them by a factor of 2.  The
+   multiplications in this recurrence may require up to 48 bits of
+   precision in the result to avoid overflow.  In practice, each row of
+   the recurrence only depends on the previous row, so an implementation
+   does not need to store all of them.
+
+   silk_NLSF2A() uses the values from the last row of this recurrence to
+   reconstruct a 32-bit version of the LPC filter (without the leading
+   1.0 coefficient), a32_Q17[k], 0 <= k < d2:
+
+        a32_Q17[k]         = -(q_Q16[d2-1][k+1] - q_Q16[d2-1][k])
+                             - (p_Q16[d2-1][k+1] + p_Q16[d2-1][k]))
+
+        a32_Q17[d_LPC-k-1] =  (q_Q16[d2-1][k+1] - q_Q16[d2-1][k])
+                             - (p_Q16[d2-1][k+1] + p_Q16[d2-1][k]))
+
+   The sum and difference of two terms from each of the p_Q16 and q_Q16
+   coefficient lists reflect the (1 + z**-1) and (1 - z**-1) factors of
+   P and Q, respectively.  The promotion of the expression from Q16 to
+   Q17 implicitly scales the result by 1/2.
+
+4.2.7.5.7.  Limiting the Range of the LPC Coefficients
+
+   The a32_Q17[] coefficients are too large to fit in a 16-bit value,
+   which significantly increases the cost of applying this filter in
+   fixed-point decoders.  Reducing them to Q12 precision doesn't incur
+   any significant quality loss, but still does not guarantee they will
+   fit. silk_NLSF2A() applies up to 10 rounds of bandwidth expansion to
+   limit the dynamic range of these coefficients.  Even floating-point
+   decoders SHOULD perform these steps, to avoid mismatch.
+
+   For each round, the process first finds the index k such that
+   abs(a32_Q17[k]) is largest, breaking ties by choosing the lowest
+   value of k.  Then, it computes the corresponding Q12 precision value,
+   maxabs_Q12, subject to an upper bound to avoid overflow in subsequent
+   computations:
+
+              maxabs_Q12 = min((maxabs_Q17 + 16) >> 5, 163838)
+
+   If this is larger than 32767, the procedure derives the chirp factor,
+   sc_Q16[0], to use in the bandwidth expansion as
+
+                                   (maxabs_Q12 - 32767) << 14
+               sc_Q16[0] = 65470 - --------------------------
+                                   (maxabs_Q12 * (k+1)) >> 2
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 70]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   where the division here is integer division.  This is an
+   approximation of the chirp factor needed to reduce the target
+   coefficient to 32767, though it is both less than 0.999 and, for
+   k > 0 when maxabs_Q12 is much greater than 32767, still slightly too
+   large.  The upper bound on maxabs_Q12, 163838, was chosen because it
+   is equal to ((2**31 - 1) >> 14) + 32767, i.e., the largest value of
+   maxabs_Q12 that would not overflow the numerator in the equation
+   above when stored in a signed 32-bit integer.
+
+   silk_bwexpander_32() (bwexpander_32.c) performs the bandwidth
+   expansion (again, only when maxabs_Q12 is greater than 32767) using
+   the following recurrence:
+
+              a32_Q17[k] = (a32_Q17[k]*sc_Q16[k]) >> 16
+
+             sc_Q16[k+1] = (sc_Q16[0]*sc_Q16[k] + 32768) >> 16
+
+   The first multiply may require up to 48 bits of precision in the
+   result to avoid overflow.  The second multiply must be unsigned to
+   avoid overflow with only 32 bits of precision.  The reference
+   implementation uses a slightly more complex formulation that avoids
+   the 32-bit overflow using signed multiplication, but is otherwise
+   equivalent.
+
+   After 10 rounds of bandwidth expansion are performed, they are simply
+   saturated to 16 bits:
+
+       a32_Q17[k] = clamp(-32768, (a32_Q17[k] + 16) >> 5, 32767) << 5
+
+   Because this performs the actual saturation in the Q12 domain, but
+   converts the coefficients back to the Q17 domain for the purposes of
+   prediction gain limiting, this step must be performed after the 10th
+   round of bandwidth expansion, regardless of whether or not the Q12
+   version of any coefficient still overflows a 16-bit integer.  This
+   saturation is not performed if maxabs_Q12 drops to 32767 or less
+   prior to the 10th round.
+
+4.2.7.5.8.  Limiting the Prediction Gain of the LPC Filter
+
+   The prediction gain of an LPC synthesis filter is the square root of
+   the output energy when the filter is excited by a unit-energy
+   impulse.  Even if the Q12 coefficients would fit, the resulting
+   filter may still have a significant gain (especially for voiced
+   sounds), making the filter unstable. silk_NLSF2A() applies up to 16
+   additional rounds of bandwidth expansion to limit the prediction
+   gain.  Instead of controlling the amount of bandwidth expansion using
+   the prediction gain itself (which may diverge to infinity for an
+   unstable filter), silk_NLSF2A() uses silk_LPC_inverse_pred_gain_QA()
+
+
+
+Valin, et al.                Standards Track                   [Page 71]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   (LPC_inv_pred_gain.c) to compute the reflection coefficients
+   associated with the filter.  The filter is stable if and only if the
+   magnitude of these coefficients is sufficiently less than one.  The
+   reflection coefficients, rc[k], can be computed using a simple
+   Levinson recurrence, initialized with the LPC coefficients a[d_LPC-
+   1][n] = a[n], and then updated via
+
+                      rc[k] = -a[k][k] ,
+
+                              a[k][n] - a[k][k-n-1]*rc[k]
+                  a[k-1][n] = ---------------------------
+                                               2
+                                      1 - rc[k]
+
+   However, silk_LPC_inverse_pred_gain_QA() approximates this using
+   fixed-point arithmetic to guarantee reproducible results across
+   platforms and implementations.  Since small changes in the
+   coefficients can make a stable filter unstable, it takes the real Q12
+   coefficients that will be used during reconstruction as input.  Thus,
+   let
+
+                    a32_Q12[n] = (a32_Q17[n] + 16) >> 5
+
+   be the Q12 version of the LPC coefficients that will eventually be
+   used.  As a simple initial check, the decoder computes the DC
+   response as
+
+                                  d_PLC-1
+                                    __
+                          DC_resp = \   a32_Q12[n]
+                                    /_
+                                    n=0
+
+   and if DC_resp > 4096, the filter is unstable.
+
+   Increasing the precision of these Q12 coefficients to Q24 for
+   intermediate computations allows more accurate computation of the
+   reflection coefficients, so the decoder initializes the recurrence
+   via
+
+                   inv_gain_Q30[d_LPC] = 1 << 30
+
+                   a32_Q24[d_LPC-1][n] = a32_Q12[n] << 12
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 72]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Then, for each k from d_LPC-1 down to 0, if
+   abs(a32_Q24[k][k]) > 16773022, the filter is unstable and the
+   recurrence stops.  The constant 16773022 here is approximately
+   0.99975 in Q24.  Otherwise, the inverse of the prediction gain,
+   inv_gain_Q30[k], is updated via
+
+              rc_Q31[k] = -a32_Q24[k][k] << 7
+
+             div_Q30[k] = (1<<30) - (rc_Q31[k]*rc_Q31[k] >> 32)
+
+        inv_gain_Q30[k] = (inv_gain_Q30[k+1]*div_Q30[k] >> 32) << 2
+
+   and if inv_gain_Q30[k] < 107374, the filter is unstable and the
+   recurrence stops.  The constant 107374 here is approximately 1/10000
+   in Q30.  If neither of these checks determine that the filter is
+   unstable and k > 0, row k-1 of a32_Q24 is computed from row k as
+
+              b1[k] = ilog(div_Q30[k])
+
+              b2[k] = b1[k] - 16
+
+                            (1<<29) - 1
+         inv_Qb2[k] = -----------------------
+                      div_Q30[k] >> (b2[k]+1)
+
+         err_Q29[k] = (1<<29)
+                      - ((div_Q30[k]<<(15-b2[k]))*inv_Qb2[k] >> 16)
+
+        gain_Qb1[k] = ((inv_Qb2[k] << 16)
+                       + (err_Q29[k]*inv_Qb2[k] >> 13))
+
+    num_Q24[k-1][n] = a32_Q24[k][n]
+                      - ((a32_Q24[k][k-n-1]*rc_Q31[k] + (1<<30)) >> 31)
+
+    a32_Q24[k-1][n] = (num_Q24[k-1][n]*gain_Qb1[k]
+                       + (1<<(b1[k]-1))) >> b1[k]
+
+   where 0 <= n < k.  In the above, rc_Q31[k] are the reflection
+   coefficients. div_Q30[k] is the denominator for each iteration, and
+   gain_Qb1[k] is its multiplicative inverse (with b1[k] fractional
+   bits, where b1[k] ranges from 20 to 31). inv_Qb2[k], which ranges
+   from 16384 to 32767, is a low-precision version of that inverse (with
+   b2[k] fractional bits). err_Q29[k] is the residual error, ranging
+   from -32763 to 32392, which is used to improve the accuracy.  The
+   values t_Q24[k-1][n] for each n are the numerators for the next row
+   of coefficients in the recursion, and a32_Q24[k-1][n] is the final
+   version of that row.  Every multiply in this procedure except the one
+   used to compute gain_Qb1[k] requires more than 32 bits of precision,
+
+
+
+Valin, et al.                Standards Track                   [Page 73]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   but otherwise all intermediate results fit in 32 bits or less.  In
+   practice, because each row only depends on the next one, an
+   implementation does not need to store them all.
+
+   If abs(a32_Q24[k][k]) <= 16773022 and inv_gain_Q30[k] >= 107374 for
+   0 <= k < d_LPC, then the filter is considered stable.  However, the
+   problem of determining stability is ill-conditioned when the filter
+   contains several reflection coefficients whose magnitude is very
+   close to one.  This fixed-point algorithm is not mathematically
+   guaranteed to correctly classify filters as stable or unstable in
+   this case, though it does very well in practice.
+
+   On round i, 0 <= i < 16, if the filter passes these stability checks,
+   then this procedure stops, and the final LPC coefficients to use for
+   reconstruction in Section 4.2.7.9.2 are
+
+                     a_Q12[k] = (a32_Q17[k] + 16) >> 5
+
+   Otherwise, a round of bandwidth expansion is applied using the same
+   procedure as in Section 4.2.7.5.7, with
+
+                         sc_Q16[0] = 65536 - (2<<i)
+
+   During round 15, sc_Q16[0] becomes 0 in the above equation, so
+   a_Q12[k] is set to 0 for all k, guaranteeing a stable filter.
+
+4.2.7.6.  Long-Term Prediction (LTP) Parameters
+
+   After the normalized LSF indices and, for 20 ms frames, the LSF
+   interpolation index, voiced frames (see Section 4.2.7.3) include
+   additional LTP parameters.  There is one primary lag index for each
+   SILK frame, but this is refined to produce a separate lag index per
+   subframe using a vector quantizer.  Each subframe also gets its own
+   prediction gain coefficient.
+
+4.2.7.6.1.  Pitch Lags
+
+   The primary lag index is coded either relative to the primary lag of
+   the prior frame in the same channel or as an absolute index.
+   Absolute coding is used if and only if
+
+   o  This is the first SILK frame of its type (LBRR or regular) for
+      this channel in the current Opus frame,
+
+   o  The previous SILK frame of the same type (LBRR or regular) for
+      this channel in the same Opus frame was not coded, or
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 74]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  That previous SILK frame was coded, but was not voiced (see
+      Section 4.2.7.3).
+
+   With absolute coding, the primary pitch lag may range from 2 ms
+   (inclusive) up to 18 ms (exclusive), corresponding to pitches from
+   500 Hz down to 55.6 Hz, respectively.  It is comprised of a high part
+   and a low part, where the decoder first reads the high part using the
+   32-entry codebook in Table 29 and then the low part using the
+   codebook corresponding to the current audio bandwidth from Table 30.
+   The final primary pitch lag is then
+
+                lag = lag_high*lag_scale + lag_low + lag_min
+
+   where lag_high is the high part, lag_low is the low part, and
+   lag_scale and lag_min are the values from the "Scale" and "Minimum
+   Lag" columns of Table 30, respectively.
+
+   +-------------------------------------------------------------------+
+   | PDF                                                               |
+   +-------------------------------------------------------------------+
+   | {3, 3, 6, 11, 21, 30, 32, 19, 11, 10, 12, 13, 13, 12, 11, 9, 8,   |
+   | 7, 6, 4, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1}/256                  |
+   +-------------------------------------------------------------------+
+
+             Table 29: PDF for High Part of Primary Pitch Lag
+
+   +------------+------------------------+-------+----------+----------+
+   | Audio      | PDF                    | Scale | Minimum  | Maximum  |
+   | Bandwidth  |                        |       | Lag      | Lag      |
+   +------------+------------------------+-------+----------+----------+
+   | NB         | {64, 64, 64, 64}/256   | 4     | 16       | 144      |
+   |            |                        |       |          |          |
+   | MB         | {43, 42, 43, 43, 42,   | 6     | 24       | 216      |
+   |            | 43}/256                |       |          |          |
+   |            |                        |       |          |          |
+   | WB         | {32, 32, 32, 32, 32,   | 8     | 32       | 288      |
+   |            | 32, 32, 32}/256        |       |          |          |
+   +------------+------------------------+-------+----------+----------+
+
+              Table 30: PDF for Low Part of Primary Pitch Lag
+
+   All frames that do not use absolute coding for the primary lag index
+   use relative coding instead.  The decoder reads a single delta value
+   using the 21-entry PDF in Table 31.  If the resulting value is zero,
+   it falls back to the absolute coding procedure from the prior
+   paragraph.  Otherwise, the final primary pitch lag is then
+
+                 lag = previous_lag + (delta_lag_index - 9)
+
+
+
+Valin, et al.                Standards Track                   [Page 75]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   where previous_lag is the primary pitch lag from the most recent
+   frame in the same channel and delta_lag_index is the value just
+   decoded.  This allows a per-frame change in the pitch lag of -8 to
+   +11 samples.  The decoder does no clamping at this point, so this
+   value can fall outside the range of 2 ms to 18 ms, and the decoder
+   must use this unclamped value when using relative coding in the next
+   SILK frame (if any).  However, because an Opus frame can use relative
+   coding for at most two consecutive SILK frames, integer overflow
+   should not be an issue.
+
+   +-------------------------------------------------------------------+
+   | PDF                                                               |
+   +-------------------------------------------------------------------+
+   | {46, 2, 2, 3, 4, 6, 10, 15, 26, 38, 30, 22, 15, 10, 7, 6, 4, 4,   |
+   | 2, 2, 2}/256                                                      |
+   +-------------------------------------------------------------------+
+
+                Table 31: PDF for Primary Pitch Lag Change
+
+   After the primary pitch lag, a "pitch contour", stored as a single
+   entry from one of four small VQ codebooks, gives lag offsets for each
+   subframe in the current SILK frame.  The codebook index is decoded
+   using one of the PDFs in Table 32 depending on the current frame size
+   and audio bandwidth.  Tables 33 through 36 give the corresponding
+   offsets to apply to the primary pitch lag for each subframe given the
+   decoded codebook index.
+
+   +-----------+--------+----------+-----------------------------------+
+   | Audio     | SILK   | Codebook | PDF                               |
+   | Bandwidth | Frame  |     Size |                                   |
+   |           | Size   |          |                                   |
+   +-----------+--------+----------+-----------------------------------+
+   | NB        | 10 ms  |        3 | {143, 50, 63}/256                 |
+   |           |        |          |                                   |
+   | NB        | 20 ms  |       11 | {68, 12, 21, 17, 19, 22, 30, 24,  |
+   |           |        |          | 17, 16, 10}/256                   |
+   |           |        |          |                                   |
+   | MB or WB  | 10 ms  |       12 | {91, 46, 39, 19, 14, 12, 8, 7, 6, |
+   |           |        |          | 5, 5, 4}/256                      |
+   |           |        |          |                                   |
+   | MB or WB  | 20 ms  |       34 | {33, 22, 18, 16, 15, 14, 14, 13,  |
+   |           |        |          | 13, 10, 9, 9, 8, 6, 6, 6, 5, 4,   |
+   |           |        |          | 4, 4, 3, 3, 3, 2, 2, 2, 2, 2, 2,  |
+   |           |        |          | 2, 1, 1, 1, 1}/256                |
+   +-----------+--------+----------+-----------------------------------+
+
+                 Table 32: PDFs for Subframe Pitch Contour
+
+
+
+
+Valin, et al.                Standards Track                   [Page 76]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       +-------+------------------+
+                       | Index | Subframe Offsets |
+                       +-------+------------------+
+                       | 0     |             0  0 |
+                       |       |                  |
+                       | 1     |             1  0 |
+                       |       |                  |
+                       | 2     |             0  1 |
+                       +-------+------------------+
+
+          Table 33: Codebook Vectors for Subframe Pitch Contour:
+                             NB, 10 ms Frames
+
+                       +-------+------------------+
+                       | Index | Subframe Offsets |
+                       +-------+------------------+
+                       | 0     |       0  0  0  0 |
+                       |       |                  |
+                       | 1     |       2  1  0 -1 |
+                       |       |                  |
+                       | 2     |      -1  0  1  2 |
+                       |       |                  |
+                       | 3     |      -1  0  0  1 |
+                       |       |                  |
+                       | 4     |      -1  0  0  0 |
+                       |       |                  |
+                       | 5     |       0  0  0  1 |
+                       |       |                  |
+                       | 6     |       0  0  1  1 |
+                       |       |                  |
+                       | 7     |       1  1  0  0 |
+                       |       |                  |
+                       | 8     |       1  0  0  0 |
+                       |       |                  |
+                       | 9     |       0  0  0 -1 |
+                       |       |                  |
+                       | 10    |       1  0  0 -1 |
+                       +-------+------------------+
+
+          Table 34: Codebook Vectors for Subframe Pitch Contour:
+                             NB, 20 ms Frames
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 77]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       +-------+------------------+
+                       | Index | Subframe Offsets |
+                       +-------+------------------+
+                       | 0     |             0  0 |
+                       |       |                  |
+                       | 1     |             0  1 |
+                       |       |                  |
+                       | 2     |             1  0 |
+                       |       |                  |
+                       | 3     |            -1  1 |
+                       |       |                  |
+                       | 4     |             1 -1 |
+                       |       |                  |
+                       | 5     |            -1  2 |
+                       |       |                  |
+                       | 6     |             2 -1 |
+                       |       |                  |
+                       | 7     |            -2  2 |
+                       |       |                  |
+                       | 8     |             2 -2 |
+                       |       |                  |
+                       | 9     |            -2  3 |
+                       |       |                  |
+                       | 10    |             3 -2 |
+                       |       |                  |
+                       | 11    |            -3  3 |
+                       +-------+------------------+
+
+     Table 35: Codebook Vectors for Subframe Pitch Contour: MB or WB,
+                               10 ms Frames
+
+                       +-------+------------------+
+                       | Index | Subframe Offsets |
+                       +-------+------------------+
+                       | 0     |       0  0  0  0 |
+                       |       |                  |
+                       | 1     |       0  0  1  1 |
+                       |       |                  |
+                       | 2     |       1  1  0  0 |
+                       |       |                  |
+                       | 3     |      -1  0  0  0 |
+                       |       |                  |
+                       | 4     |       0  0  0  1 |
+                       |       |                  |
+                       | 5     |       1  0  0  0 |
+                       |       |                  |
+                       | 6     |      -1  0  0  1 |
+                       |       |                  |
+
+
+
+Valin, et al.                Standards Track                   [Page 78]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       | 7     |       0  0  0 -1 |
+                       |       |                  |
+                       | 8     |      -1  0  1  2 |
+                       |       |                  |
+                       | 9     |       1  0  0 -1 |
+                       |       |                  |
+                       | 10    |      -2 -1  1  2 |
+                       |       |                  |
+                       | 11    |       2  1  0 -1 |
+                       |       |                  |
+                       | 12    |      -2  0  0  2 |
+                       |       |                  |
+                       | 13    |      -2  0  1  3 |
+                       |       |                  |
+                       | 14    |       2  1 -1 -2 |
+                       |       |                  |
+                       | 15    |      -3 -1  1  3 |
+                       |       |                  |
+                       | 16    |       2  0  0 -2 |
+                       |       |                  |
+                       | 17    |       3  1  0 -2 |
+                       |       |                  |
+                       | 18    |      -3 -1  2  4 |
+                       |       |                  |
+                       | 19    |      -4 -1  1  4 |
+                       |       |                  |
+                       | 20    |       3  1 -1 -3 |
+                       |       |                  |
+                       | 21    |      -4 -1  2  5 |
+                       |       |                  |
+                       | 22    |       4  2 -1 -3 |
+                       |       |                  |
+                       | 23    |       4  1 -1 -4 |
+                       |       |                  |
+                       | 24    |      -5 -1  2  6 |
+                       |       |                  |
+                       | 25    |       5  2 -1 -4 |
+                       |       |                  |
+                       | 26    |      -6 -2  2  6 |
+                       |       |                  |
+                       | 27    |      -5 -2  2  5 |
+                       |       |                  |
+                       | 28    |       6  2 -1 -5 |
+                       |       |                  |
+                       | 29    |      -7 -2  3  8 |
+                       |       |                  |
+                       | 30    |       6  2 -2 -6 |
+                       |       |                  |
+
+
+
+Valin, et al.                Standards Track                   [Page 79]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       | 31    |       5  2 -2 -5 |
+                       |       |                  |
+                       | 32    |       8  3 -2 -7 |
+                       |       |                  |
+                       | 33    |      -9 -3  3  9 |
+                       +-------+------------------+
+
+     Table 36: Codebook Vectors for Subframe Pitch Contour: MB or WB,
+                               20 ms Frames
+
+   The final pitch lag for each subframe is assembled in
+   silk_decode_pitch() (decode_pitch.c).  Let lag be the primary pitch
+   lag for the current SILK frame, contour_index be index of the VQ
+   codebook, and lag_cb[contour_index][k] be the corresponding entry of
+   the codebook from the appropriate table given above for the k'th
+   subframe.  Then the final pitch lag for that subframe is
+
+       pitch_lags[k] = clamp(lag_min, lag + lag_cb[contour_index][k],
+                             lag_max)
+
+   where lag_min and lag_max are the values from the "Minimum Lag" and
+   "Maximum Lag" columns of Table 30, respectively.
+
+4.2.7.6.2.  LTP Filter Coefficients
+
+   SILK uses a separate 5-tap pitch filter for each subframe, selected
+   from one of three codebooks.  The three codebooks each represent
+   different rate-distortion trade-offs, with average rates of
+   1.61 bits/subframe, 3.68 bits/subframe, and 4.85 bits/subframe,
+   respectively.
+
+   The importance of the filter coefficients generally depends on two
+   factors: the periodicity of the signal and relative energy between
+   the current subframe and the signal from one period earlier.  Greater
+   periodicity and decaying energy both lead to more important filter
+   coefficients.  Thus, they should be coded with lower distortion and
+   higher rate.  These properties are relatively stable over the
+   duration of a single SILK frame.  Hence, all of the subframes in a
+   SILK frame choose their filter from the same codebook.  This is
+   signaled with an explicitly-coded "periodicity index".  This
+   immediately follows the subframe pitch lags, and is coded using the
+   3-entry PDF from Table 37.
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 80]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                           +------------------+
+                           | PDF              |
+                           +------------------+
+                           | {77, 80, 99}/256 |
+                           +------------------+
+
+                      Table 37: Periodicity Index PDF
+
+   The indices of the filters for each subframe follow.  They are all
+   coded using the PDF from Table 38 corresponding to the periodicity
+   index.  Tables 39 through 41 contain the corresponding filter taps as
+   signed Q7 integers.
+
+   +-------------+----------+------------------------------------------+
+   | Periodicity | Codebook | PDF                                      |
+   | Index       |     Size |                                          |
+   +-------------+----------+------------------------------------------+
+   | 0           |        8 | {185, 15, 13, 13, 9, 9, 6, 6}/256        |
+   |             |          |                                          |
+   | 1           |       16 | {57, 34, 21, 20, 15, 13, 12, 13, 10, 10, |
+   |             |          | 9, 10, 9, 8, 7, 8}/256                   |
+   |             |          |                                          |
+   | 2           |       32 | {15, 16, 14, 12, 12, 12, 11, 11, 11, 10, |
+   |             |          | 9, 9, 9, 9, 8, 8, 8, 8, 7, 7, 6, 6, 5,   |
+   |             |          | 4, 5, 4, 4, 4, 3, 4, 3, 2}/256           |
+   +-------------+----------+------------------------------------------+
+
+                         Table 38: LTP Filter PDFs
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 81]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                      +-------+---------------------+
+                      | Index |    Filter Taps (Q7) |
+                      +-------+---------------------+
+                      | 0     |   4   6  24   7   5 |
+                      |       |                     |
+                      | 1     |   0   0   2   0   0 |
+                      |       |                     |
+                      | 2     |  12  28  41  13  -4 |
+                      |       |                     |
+                      | 3     |  -9  15  42  25  14 |
+                      |       |                     |
+                      | 4     |   1  -2  62  41  -9 |
+                      |       |                     |
+                      | 5     | -10  37  65  -4   3 |
+                      |       |                     |
+                      | 6     |  -6   4  66   7  -8 |
+                      |       |                     |
+                      | 7     |  16  14  38  -3  33 |
+                      +-------+---------------------+
+
+      Table 39: Codebook Vectors for LTP Filter, Periodicity Index 0
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 82]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                      +-------+---------------------+
+                      | Index |    Filter Taps (Q7) |
+                      +-------+---------------------+
+                      | 0     |  13  22  39  23  12 |
+                      |       |                     |
+                      | 1     |  -1  36  64  27  -6 |
+                      |       |                     |
+                      | 2     |  -7  10  55  43  17 |
+                      |       |                     |
+                      | 3     |   1   1   8   1   1 |
+                      |       |                     |
+                      | 4     |   6 -11  74  53  -9 |
+                      |       |                     |
+                      | 5     | -12  55  76 -12   8 |
+                      |       |                     |
+                      | 6     |  -3   3  93  27  -4 |
+                      |       |                     |
+                      | 7     |  26  39  59   3  -8 |
+                      |       |                     |
+                      | 8     |   2   0  77  11   9 |
+                      |       |                     |
+                      | 9     |  -8  22  44  -6   7 |
+                      |       |                     |
+                      | 10    |  40   9  26   3   9 |
+                      |       |                     |
+                      | 11    |  -7  20 101  -7   4 |
+                      |       |                     |
+                      | 12    |   3  -8  42  26   0 |
+                      |       |                     |
+                      | 13    | -15  33  68   2  23 |
+                      |       |                     |
+                      | 14    |  -2  55  46  -2  15 |
+                      |       |                     |
+                      | 15    |   3  -1  21  16  41 |
+                      +-------+---------------------+
+
+      Table 40: Codebook Vectors for LTP Filter, Periodicity Index 1
+
+                      +-------+---------------------+
+                      | Index |    Filter Taps (Q7) |
+                      +-------+---------------------+
+                      | 0     |  -6  27  61  39   5 |
+                      |       |                     |
+                      | 1     | -11  42  88   4   1 |
+                      |       |                     |
+                      | 2     |  -2  60  65   6  -4 |
+                      |       |                     |
+                      | 3     |  -1  -5  73  56   1 |
+
+
+
+Valin, et al.                Standards Track                   [Page 83]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                      | 4     |  -9  19  94  29  -9 |
+                      |       |                     |
+                      | 5     |   0  12  99   6   4 |
+                      |       |                     |
+                      | 6     |   8 -19 102  46 -13 |
+                      |       |                     |
+                      | 7     |   3   2  13   3   2 |
+                      |       |                     |
+                      | 8     |   9 -21  84  72 -18 |
+                      |       |                     |
+                      | 9     | -11  46 104 -22   8 |
+                      |       |                     |
+                      | 10    |  18  38  48  23   0 |
+                      |       |                     |
+                      | 11    | -16  70  83 -21  11 |
+                      |       |                     |
+                      | 12    |   5 -11 117  22  -8 |
+                      |       |                     |
+                      | 13    |  -6  23 117 -12   3 |
+                      |       |                     |
+                      | 14    |   3  -8  95  28   4 |
+                      |       |                     |
+                      | 15    | -10  15  77  60 -15 |
+                      |       |                     |
+                      | 16    |  -1   4 124   2  -4 |
+                      |       |                     |
+                      | 17    |   3  38  84  24 -25 |
+                      |       |                     |
+                      | 18    |   2  13  42  13  31 |
+                      |       |                     |
+                      | 19    |  21  -4  56  46  -1 |
+                      |       |                     |
+                      | 20    |  -1  35  79 -13  19 |
+                      |       |                     |
+                      | 21    |  -7  65  88  -9 -14 |
+                      |       |                     |
+                      | 22    |  20   4  81  49 -29 |
+                      |       |                     |
+                      | 23    |  20   0  75   3 -17 |
+                      |       |                     |
+                      | 24    |   5  -9  44  92  -8 |
+                      |       |                     |
+                      | 25    |   1  -3  22  69  31 |
+                      |       |                     |
+                      | 26    |  -6  95  41 -12   5 |
+                      |       |                     |
+                      | 27    |  39  67  16  -4   1 |
+                      |       |                     |
+
+
+
+Valin, et al.                Standards Track                   [Page 84]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                      | 28    |   0  -6 120  55 -36 |
+                      |       |                     |
+                      | 29    | -13  44 122   4 -24 |
+                      |       |                     |
+                      | 30    |  81   5  11   3   7 |
+                      |       |                     |
+                      | 31    |   2   0   9  10  88 |
+                      +-------+---------------------+
+
+      Table 41: Codebook Vectors for LTP Filter, Periodicity Index 2
+
+4.2.7.6.3.  LTP Scaling Parameter
+
+   An LTP scaling parameter appears after the LTP filter coefficients if
+   and only if
+
+   o  This is a voiced frame (see Section 4.2.7.3), and
+
+   o  Either
+
+      *  This SILK frame corresponds to the first time interval of the
+         current Opus frame for its type (LBRR or regular), or
+
+      *  This is an LBRR frame where the LBRR flags (see Section 4.2.4)
+         indicate the previous LBRR frame in the same channel is not
+         coded.
+
+   This allows the encoder to trade off the prediction gain between
+   packets against the recovery time after packet loss.  Unlike
+   absolute-coding for pitch lags, regular SILK frames that are not at
+   the start of an Opus frame (i.e., that do not correspond to the first
+   20 ms time interval in Opus frames of 40 or 60 ms) do not include
+   this field, even if the prior frame was not voiced, or (in the case
+   of the side channel) not even coded.  After an uncoded frame in the
+   side channel, the LTP buffer (see Section 4.2.7.9.1) is cleared to
+   zero, and is thus in a known state.  In contrast, LBRR frames do
+   include this field when the prior frame was not coded, since the LTP
+   buffer contains the output of the PLC, which is non-normative.
+
+   If present, the decoder reads a value using the 3-entry PDF in
+   Table 42.  The three possible values represent Q14 scale factors of
+   15565, 12288, and 8192, respectively (corresponding to approximately
+   0.95, 0.75, and 0.5).  Frames that do not code the scaling parameter
+   use the default factor of 15565 (approximately 0.95).
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 85]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                           +-------------------+
+                           | PDF               |
+                           +-------------------+
+                           | {128, 64, 64}/256 |
+                           +-------------------+
+
+                  Table 42: PDF for LTP Scaling Parameter
+
+4.2.7.7.  Linear Congruential Generator (LCG) Seed
+
+   As described in Section 4.2.7.8.6, SILK uses a Linear Congruential
+   Generator (LCG) to inject pseudorandom noise into the quantized
+   excitation.  To ensure synchronization of this process between the
+   encoder and decoder, each SILK frame stores a 2-bit seed after the
+   LTP parameters (if any).  The encoder may consider the choice of seed
+   during quantization, and the flexibility of this choice lets it
+   reduce distortion, helping to pay for the bit cost required to signal
+   it.  The decoder reads the seed using the uniform 4-entry PDF in
+   Table 43, yielding a value between 0 and 3, inclusive.
+
+                         +----------------------+
+                         | PDF                  |
+                         +----------------------+
+                         | {64, 64, 64, 64}/256 |
+                         +----------------------+
+
+                        Table 43: PDF for LCG Seed
+
+4.2.7.8.  Excitation
+
+   SILK codes the excitation using a modified version of the Pyramid
+   Vector Quantizer (PVQ) codebook [PVQ].  The PVQ codebook is designed
+   for Laplace-distributed values and consists of all sums of K signed,
+   unit pulses in a vector of dimension N, where two pulses at the same
+   position are required to have the same sign.  Thus, the codebook
+   includes all integer codevectors y of dimension N that satisfy
+
+                              N-1
+                              __
+                              \  abs(y[j]) = K
+                              /_
+                              j=0
+
+   Unlike regular PVQ, SILK uses a variable-length, rather than fixed-
+   length, encoding.  This encoding is better suited to the more
+   Gaussian-like distribution of the coefficient magnitudes and the non-
+   uniform distribution of their signs (caused by the quantization
+   offset described below).  SILK also handles large codebooks by coding
+
+
+
+Valin, et al.                Standards Track                   [Page 86]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   the least significant bits (LSBs) of each coefficient directly.  This
+   adds a small coding efficiency loss, but greatly reduces the
+   computation time and ROM size required for decoding, as implemented
+   in silk_decode_pulses() (decode_pulses.c).
+
+   SILK fixes the dimension of the codebook to N = 16.  The excitation
+   is made up of a number of "shell blocks", each 16 samples in size.
+   Table 44 lists the number of shell blocks required for a SILK frame
+   for each possible audio bandwidth and frame size. 10 ms MB frames
+   nominally contain 120 samples (10 ms at 12 kHz), which is not a
+   multiple of 16.  This is handled by coding 8 shell blocks (128
+   samples) and discarding the final 8 samples of the last block.  The
+   decoder contains no special case that prevents an encoder from
+   placing pulses in these samples, and they must be correctly parsed
+   from the bitstream if present, but they are otherwise ignored.
+
+         +-----------------+------------+------------------------+
+         | Audio Bandwidth | Frame Size | Number of Shell Blocks |
+         +-----------------+------------+------------------------+
+         | NB              | 10 ms      |                      5 |
+         |                 |            |                        |
+         | MB              | 10 ms      |                      8 |
+         |                 |            |                        |
+         | WB              | 10 ms      |                     10 |
+         |                 |            |                        |
+         | NB              | 20 ms      |                     10 |
+         |                 |            |                        |
+         | MB              | 20 ms      |                     15 |
+         |                 |            |                        |
+         | WB              | 20 ms      |                     20 |
+         +-----------------+------------+------------------------+
+
+              Table 44: Number of Shell Blocks Per SILK Frame
+
+4.2.7.8.1.  Rate Level
+
+   The first symbol in the excitation is a "rate level", which is an
+   index from 0 to 8, inclusive, coded using the PDF in Table 45
+   corresponding to the signal type of the current frame (from
+   Section 4.2.7.3).  The rate level selects the PDF used to decode the
+   number of pulses in the individual shell blocks.  It does not
+   directly convey any information about the bitrate or the number of
+   pulses itself, but merely changes the probability of the symbols in
+   Section 4.2.7.8.2.  Level 0 provides a more efficient encoding at low
+   rates generally, and level 8 provides a more efficient encoding at
+   high rates generally, though the most efficient level for a
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 87]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   particular SILK frame may depend on the exact distribution of the
+   coded symbols.  An encoder should, but is not required to, use the
+   most efficient rate level.
+
+    +----------------------+------------------------------------------+
+    | Signal Type          | PDF                                      |
+    +----------------------+------------------------------------------+
+    | Inactive or Unvoiced | {15, 51, 12, 46, 45, 13, 33, 27, 14}/256 |
+    |                      |                                          |
+    | Voiced               | {33, 30, 36, 17, 34, 49, 18, 21, 18}/256 |
+    +----------------------+------------------------------------------+
+
+                     Table 45: PDFs for the Rate Level
+
+4.2.7.8.2.  Pulses per Shell Block
+
+   The total number of pulses in each of the shell blocks follows the
+   rate level.  The pulse counts for all of the shell blocks are coded
+   consecutively, before the content of any of the blocks.  Each block
+   may have anywhere from 0 to 16 pulses, inclusive, coded using the 18-
+   entry PDF in Table 46 corresponding to the rate level from
+   Section 4.2.7.8.1.  The special value 17 indicates that this block
+   has one or more additional LSBs to decode for each coefficient.  If
+   the decoder encounters this value, it decodes another value for the
+   actual pulse count of the block, but uses the PDF corresponding to
+   the special rate level 9 instead of the normal rate level.  This
+   process repeats until the decoder reads a value less than 17, and it
+   then sets the number of extra LSBs used to the number of 17's decoded
+   for that block.  If it reads the value 17 ten times, then the next
+   iteration uses the special rate level 10 instead of 9.  The
+   probability of decoding a 17 when using the PDF for rate level 10 is
+   zero, ensuring that the number of LSBs for a block will not exceed
+   10.  The cumulative distribution for rate level 10 is just a shifted
+   version of that for 9 and thus does not require any additional
+   storage.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 88]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +----------+--------------------------------------------------------+
+   | Rate     | PDF                                                    |
+   | Level    |                                                        |
+   +----------+--------------------------------------------------------+
+   | 0        | {131, 74, 25, 8, 3, 3, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,   |
+   |          | 1, 1}/256                                              |
+   |          |                                                        |
+   | 1        | {58, 93, 60, 23, 7, 3, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,   |
+   |          | 1, 1}/256                                              |
+   |          |                                                        |
+   | 2        | {43, 51, 46, 33, 24, 16, 11, 8, 6, 3, 3, 3, 2, 1, 1,   |
+   |          | 2, 1, 2}/256                                           |
+   |          |                                                        |
+   | 3        | {17, 52, 71, 57, 31, 12, 5, 1, 1, 1, 1, 1, 1, 1, 1, 1, |
+   |          | 1, 1}/256                                              |
+   |          |                                                        |
+   | 4        | {6, 21, 41, 53, 49, 35, 21, 11, 6, 3, 2, 2, 1, 1, 1,   |
+   |          | 1, 1, 1}/256                                           |
+   |          |                                                        |
+   | 5        | {7, 14, 22, 28, 29, 28, 25, 20, 17, 13, 11, 9, 7, 5,   |
+   |          | 4, 4, 3, 10}/256                                       |
+   |          |                                                        |
+   | 6        | {2, 5, 14, 29, 42, 46, 41, 31, 19, 11, 6, 3, 2, 1, 1,  |
+   |          | 1, 1, 1}/256                                           |
+   |          |                                                        |
+   | 7        | {1, 2, 4, 10, 19, 29, 35, 37, 34, 28, 20, 14, 8, 5, 4, |
+   |          | 2, 2, 2}/256                                           |
+   |          |                                                        |
+   | 8        | {1, 2, 2, 5, 9, 14, 20, 24, 27, 28, 26, 23, 20, 15,    |
+   |          | 11, 8, 6, 15}/256                                      |
+   |          |                                                        |
+   | 9        | {1, 1, 1, 6, 27, 58, 56, 39, 25, 14, 10, 6, 3, 3, 2,   |
+   |          | 1, 1, 2}/256                                           |
+   |          |                                                        |
+   | 10       | {2, 1, 6, 27, 58, 56, 39, 25, 14, 10, 6, 3, 3, 2, 1,   |
+   |          | 1, 2, 0}/256                                           |
+   +----------+--------------------------------------------------------+
+
+                    Table 46: PDFs for the Pulse Count
+
+4.2.7.8.3.  Pulse Location Decoding
+
+   The locations of the pulses in each shell block follow the pulse
+   counts, as decoded by silk_shell_decoder() (shell_coder.c).  As with
+   the pulse counts, these locations are coded for all the shell blocks
+   before any of the remaining information for each block.  Unlike many
+   other codecs, SILK places no restriction on the distribution of
+
+
+
+
+Valin, et al.                Standards Track                   [Page 89]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   pulses within a shell block.  All of the pulses may be placed in a
+   single location, or each one in a unique location, or anything in
+   between.
+
+   The location of pulses is coded by recursively partitioning each
+   block into halves, and coding how many pulses fall on the left side
+   of the split.  All remaining pulses must fall on the right side of
+   the split.  The process then recurses into the left half, and after
+   that returns, the right half (preorder traversal).  The PDF to use is
+   chosen by the size of the current partition (16, 8, 4, or 2) and the
+   number of pulses in the partition (1 to 16, inclusive).  Tables 47
+   through 50 list the PDFs used for each partition size and pulse
+   count.  This process skips partitions without any pulses, i.e., where
+   the initial pulse count from Section 4.2.7.8.2 was zero, or where the
+   split in the prior level indicated that all of the pulses fell on the
+   other side.  These partitions have nothing to code, so they require
+   no PDF.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 90]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +------------+------------------------------------------------------+
+   | Pulse      | PDF                                                  |
+   | Count      |                                                      |
+   +------------+------------------------------------------------------+
+   | 1          | {126, 130}/256                                       |
+   |            |                                                      |
+   | 2          | {56, 142, 58}/256                                    |
+   |            |                                                      |
+   | 3          | {25, 101, 104, 26}/256                               |
+   |            |                                                      |
+   | 4          | {12, 60, 108, 64, 12}/256                            |
+   |            |                                                      |
+   | 5          | {7, 35, 84, 87, 37, 6}/256                           |
+   |            |                                                      |
+   | 6          | {4, 20, 59, 86, 63, 21, 3}/256                       |
+   |            |                                                      |
+   | 7          | {3, 12, 38, 72, 75, 42, 12, 2}/256                   |
+   |            |                                                      |
+   | 8          | {2, 8, 25, 54, 73, 59, 27, 7, 1}/256                 |
+   |            |                                                      |
+   | 9          | {2, 5, 17, 39, 63, 65, 42, 18, 4, 1}/256             |
+   |            |                                                      |
+   | 10         | {1, 4, 12, 28, 49, 63, 54, 30, 11, 3, 1}/256         |
+   |            |                                                      |
+   | 11         | {1, 4, 8, 20, 37, 55, 57, 41, 22, 8, 2, 1}/256       |
+   |            |                                                      |
+   | 12         | {1, 3, 7, 15, 28, 44, 53, 48, 33, 16, 6, 1, 1}/256   |
+   |            |                                                      |
+   | 13         | {1, 2, 6, 12, 21, 35, 47, 48, 40, 25, 12, 5, 1,      |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 14         | {1, 1, 4, 10, 17, 27, 37, 47, 43, 33, 21, 9, 4, 1,   |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 15         | {1, 1, 1, 8, 14, 22, 33, 40, 43, 38, 28, 16, 8, 1,   |
+   |            | 1, 1}/256                                            |
+   |            |                                                      |
+   | 16         | {1, 1, 1, 1, 13, 18, 27, 36, 41, 41, 34, 24, 14, 1,  |
+   |            | 1, 1, 1}/256                                         |
+   +------------+------------------------------------------------------+
+
+        Table 47: PDFs for Pulse Count Split, 16 Sample Partitions
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 91]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +------------+------------------------------------------------------+
+   | Pulse      | PDF                                                  |
+   | Count      |                                                      |
+   +------------+------------------------------------------------------+
+   | 1          | {127, 129}/256                                       |
+   |            |                                                      |
+   | 2          | {53, 149, 54}/256                                    |
+   |            |                                                      |
+   | 3          | {22, 105, 106, 23}/256                               |
+   |            |                                                      |
+   | 4          | {11, 61, 111, 63, 10}/256                            |
+   |            |                                                      |
+   | 5          | {6, 35, 86, 88, 36, 5}/256                           |
+   |            |                                                      |
+   | 6          | {4, 20, 59, 87, 62, 21, 3}/256                       |
+   |            |                                                      |
+   | 7          | {3, 13, 40, 71, 73, 41, 13, 2}/256                   |
+   |            |                                                      |
+   | 8          | {3, 9, 27, 53, 70, 56, 28, 9, 1}/256                 |
+   |            |                                                      |
+   | 9          | {3, 8, 19, 37, 57, 61, 44, 20, 6, 1}/256             |
+   |            |                                                      |
+   | 10         | {3, 7, 15, 28, 44, 54, 49, 33, 17, 5, 1}/256         |
+   |            |                                                      |
+   | 11         | {1, 7, 13, 22, 34, 46, 48, 38, 28, 14, 4, 1}/256     |
+   |            |                                                      |
+   | 12         | {1, 1, 11, 22, 27, 35, 42, 47, 33, 25, 10, 1, 1}/256 |
+   |            |                                                      |
+   | 13         | {1, 1, 6, 14, 26, 37, 43, 43, 37, 26, 14, 6, 1,      |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 14         | {1, 1, 4, 10, 20, 31, 40, 42, 40, 31, 20, 10, 4, 1,  |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 15         | {1, 1, 3, 8, 16, 26, 35, 38, 38, 35, 26, 16, 8, 3,   |
+   |            | 1, 1}/256                                            |
+   |            |                                                      |
+   | 16         | {1, 1, 2, 6, 12, 21, 30, 36, 38, 36, 30, 21, 12, 6,  |
+   |            | 2, 1, 1}/256                                         |
+   +------------+------------------------------------------------------+
+
+         Table 48: PDFs for Pulse Count Split, 8 Sample Partitions
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 92]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +------------+------------------------------------------------------+
+   | Pulse      | PDF                                                  |
+   | Count      |                                                      |
+   +------------+------------------------------------------------------+
+   | 1          | {127, 129}/256                                       |
+   |            |                                                      |
+   | 2          | {49, 157, 50}/256                                    |
+   |            |                                                      |
+   | 3          | {20, 107, 109, 20}/256                               |
+   |            |                                                      |
+   | 4          | {11, 60, 113, 62, 10}/256                            |
+   |            |                                                      |
+   | 5          | {7, 36, 84, 87, 36, 6}/256                           |
+   |            |                                                      |
+   | 6          | {6, 24, 57, 82, 60, 23, 4}/256                       |
+   |            |                                                      |
+   | 7          | {5, 18, 39, 64, 68, 42, 16, 4}/256                   |
+   |            |                                                      |
+   | 8          | {6, 14, 29, 47, 61, 52, 30, 14, 3}/256               |
+   |            |                                                      |
+   | 9          | {1, 15, 23, 35, 51, 50, 40, 30, 10, 1}/256           |
+   |            |                                                      |
+   | 10         | {1, 1, 21, 32, 42, 52, 46, 41, 18, 1, 1}/256         |
+   |            |                                                      |
+   | 11         | {1, 6, 16, 27, 36, 42, 42, 36, 27, 16, 6, 1}/256     |
+   |            |                                                      |
+   | 12         | {1, 5, 12, 21, 31, 38, 40, 38, 31, 21, 12, 5, 1}/256 |
+   |            |                                                      |
+   | 13         | {1, 3, 9, 17, 26, 34, 38, 38, 34, 26, 17, 9, 3,      |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 14         | {1, 3, 7, 14, 22, 29, 34, 36, 34, 29, 22, 14, 7, 3,  |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 15         | {1, 2, 5, 11, 18, 25, 31, 35, 35, 31, 25, 18, 11, 5, |
+   |            | 2, 1}/256                                            |
+   |            |                                                      |
+   | 16         | {1, 1, 4, 9, 15, 21, 28, 32, 34, 32, 28, 21, 15, 9,  |
+   |            | 4, 1, 1}/256                                         |
+   +------------+------------------------------------------------------+
+
+         Table 49: PDFs for Pulse Count Split, 4 Sample Partitions
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                   [Page 93]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +------------+------------------------------------------------------+
+   | Pulse      | PDF                                                  |
+   | Count      |                                                      |
+   +------------+------------------------------------------------------+
+   | 1          | {128, 128}/256                                       |
+   |            |                                                      |
+   | 2          | {42, 172, 42}/256                                    |
+   |            |                                                      |
+   | 3          | {21, 107, 107, 21}/256                               |
+   |            |                                                      |
+   | 4          | {12, 60, 112, 61, 11}/256                            |
+   |            |                                                      |
+   | 5          | {8, 34, 86, 86, 35, 7}/256                           |
+   |            |                                                      |
+   | 6          | {8, 23, 55, 90, 55, 20, 5}/256                       |
+   |            |                                                      |
+   | 7          | {5, 15, 38, 72, 72, 36, 15, 3}/256                   |
+   |            |                                                      |
+   | 8          | {6, 12, 27, 52, 77, 47, 20, 10, 5}/256               |
+   |            |                                                      |
+   | 9          | {6, 19, 28, 35, 40, 40, 35, 28, 19, 6}/256           |
+   |            |                                                      |
+   | 10         | {4, 14, 22, 31, 37, 40, 37, 31, 22, 14, 4}/256       |
+   |            |                                                      |
+   | 11         | {3, 10, 18, 26, 33, 38, 38, 33, 26, 18, 10, 3}/256   |
+   |            |                                                      |
+   | 12         | {2, 8, 13, 21, 29, 36, 38, 36, 29, 21, 13, 8, 2}/256 |
+   |            |                                                      |
+   | 13         | {1, 5, 10, 17, 25, 32, 38, 38, 32, 25, 17, 10, 5,    |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 14         | {1, 4, 7, 13, 21, 29, 35, 36, 35, 29, 21, 13, 7, 4,  |
+   |            | 1}/256                                               |
+   |            |                                                      |
+   | 15         | {1, 2, 5, 10, 17, 25, 32, 36, 36, 32, 25, 17, 10, 5, |
+   |            | 2, 1}/256                                            |
+   |            |                                                      |
+   | 16         | {1, 2, 4, 7, 13, 21, 28, 34, 36, 34, 28, 21, 13, 7,  |
+   |            | 4, 2, 1}/256                                         |
+   +------------+------------------------------------------------------+
+
+         Table 50: PDFs for Pulse Count Split, 2 Sample Partitions
+
+4.2.7.8.4.  LSB Decoding
+
+   After the decoder reads the pulse locations for all blocks, it reads
+   the LSBs (if any) for each block in turn.  Inside each block, it
+   reads all the LSBs for each coefficient in turn, even those where no
+
+
+
+Valin, et al.                Standards Track                   [Page 94]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   pulses were allocated, before proceeding to the next one.  For 10 ms
+   MB frames, it reads LSBs even for the extra 8 samples in the last
+   block.  The LSBs are coded from most significant to least
+   significant, and they all use the PDF in Table 51.
+
+                            +----------------+
+                            | PDF            |
+                            +----------------+
+                            | {136, 120}/256 |
+                            +----------------+
+
+                     Table 51: PDF for Excitation LSBs
+
+   The number of LSBs read for each coefficient in a block is determined
+   in Section 4.2.7.8.2.  The magnitude of the coefficient is initially
+   equal to the number of pulses placed at that location in
+   Section 4.2.7.8.3.  As each LSB is decoded, the magnitude is doubled,
+   and then the value of the LSB added to it, to obtain an updated
+   magnitude.
+
+4.2.7.8.5.  Sign Decoding
+
+   After decoding the pulse locations and the LSBs, the decoder knows
+   the magnitude of each coefficient in the excitation.  It then decodes
+   a sign for all coefficients with a non-zero magnitude, using one of
+   the PDFs from Table 52.  If the value decoded is 0, then the
+   coefficient magnitude is negated.  Otherwise, it remains positive.
+
+   The decoder chooses the PDF for the sign based on the signal type and
+   quantization offset type (from Section 4.2.7.3) and the number of
+   pulses in the block (from Section 4.2.7.8.2).  The number of pulses
+   in the block does not take into account any LSBs.  Most PDFs are
+   skewed towards negative signs because of the quantization offset, but
+   the PDFs for zero pulses are highly skewed towards positive signs.
+   If a block contains many positive coefficients, it is sometimes
+   beneficial to code it solely using LSBs (i.e., with zero pulses),
+   since the encoder may be able to save enough bits on the signs to
+   justify the less efficient coefficient magnitude encoding.
+
+   +-------------+-----------------------+-------------+---------------+
+   | Signal Type | Quantization Offset   | Pulse Count | PDF           |
+   |             | Type                  |             |               |
+   +-------------+-----------------------+-------------+---------------+
+   | Inactive    | Low                   | 0           | {2, 254}/256  |
+   |             |                       |             |               |
+   | Inactive    | Low                   | 1           | {207, 49}/256 |
+   |             |                       |             |               |
+   | Inactive    | Low                   | 2           | {189, 67}/256 |
+
+
+
+Valin, et al.                Standards Track                   [Page 95]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   | Inactive    | Low                   | 3           | {179, 77}/256 |
+   |             |                       |             |               |
+   | Inactive    | Low                   | 4           | {174, 82}/256 |
+   |             |                       |             |               |
+   | Inactive    | Low                   | 5           | {163, 93}/256 |
+   |             |                       |             |               |
+   | Inactive    | Low                   | 6 or more   | {157, 99}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 0           | {58, 198}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 1           | {245, 11}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 2           | {238, 18}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 3           | {232, 24}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 4           | {225, 31}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 5           | {220, 36}/256 |
+   |             |                       |             |               |
+   | Inactive    | High                  | 6 or more   | {211, 45}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 0           | {1, 255}/256  |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 1           | {210, 46}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 2           | {190, 66}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 3           | {178, 78}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 4           | {169, 87}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 5           | {162, 94}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | Low                   | 6 or more   | {152,         |
+   |             |                       |             | 104}/256      |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 0           | {48, 208}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 1           | {242, 14}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 2           | {235, 21}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 3           | {224, 32}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 4           | {214, 42}/256 |
+   |             |                       |             |               |
+   | Unvoiced    | High                  | 5           | {205, 51}/256 |
+
+
+
+Valin, et al.                Standards Track                   [Page 96]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   | Unvoiced    | High                  | 6 or more   | {190, 66}/256 |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 0           | {1, 255}/256  |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 1           | {162, 94}/256 |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 2           | {152,         |
+   |             |                       |             | 104}/256      |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 3           | {147,         |
+   |             |                       |             | 109}/256      |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 4           | {144,         |
+   |             |                       |             | 112}/256      |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 5           | {141,         |
+   |             |                       |             | 115}/256      |
+   |             |                       |             |               |
+   | Voiced      | Low                   | 6 or more   | {138,         |
+   |             |                       |             | 118}/256      |
+   |             |                       |             |               |
+   | Voiced      | High                  | 0           | {8, 248}/256  |
+   |             |                       |             |               |
+   | Voiced      | High                  | 1           | {203, 53}/256 |
+   |             |                       |             |               |
+   | Voiced      | High                  | 2           | {187, 69}/256 |
+   |             |                       |             |               |
+   | Voiced      | High                  | 3           | {176, 80}/256 |
+   |             |                       |             |               |
+   | Voiced      | High                  | 4           | {168, 88}/256 |
+   |             |                       |             |               |
+   | Voiced      | High                  | 5           | {161, 95}/256 |
+   |             |                       |             |               |
+   | Voiced      | High                  | 6 or more   | {154,         |
+   |             |                       |             | 102}/256      |
+   +-------------+-----------------------+-------------+---------------+
+
+                    Table 52: PDFs for Excitation Signs
+
+4.2.7.8.6.  Reconstructing the Excitation
+
+   After the signs have been read, there is enough information to
+   reconstruct the complete excitation signal.  This requires adding a
+   constant quantization offset to each non-zero sample and then
+   pseudorandomly inverting and offsetting every sample.  The constant
+   quantization offset varies depending on the signal type and
+   quantization offset type (see Section 4.2.7.3).
+
+
+
+
+Valin, et al.                Standards Track                   [Page 97]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   +-------------+--------------------------+--------------------------+
+   | Signal Type | Quantization Offset Type |      Quantization Offset |
+   |             |                          |                    (Q23) |
+   +-------------+--------------------------+--------------------------+
+   | Inactive    | Low                      |                       25 |
+   |             |                          |                          |
+   | Inactive    | High                     |                       60 |
+   |             |                          |                          |
+   | Unvoiced    | Low                      |                       25 |
+   |             |                          |                          |
+   | Unvoiced    | High                     |                       60 |
+   |             |                          |                          |
+   | Voiced      | Low                      |                        8 |
+   |             |                          |                          |
+   | Voiced      | High                     |                       25 |
+   +-------------+--------------------------+--------------------------+
+
+                 Table 53: Excitation Quantization Offsets
+
+   Let e_raw[i] be the raw excitation value at position i, with a
+   magnitude composed of the pulses at that location (see
+   Section 4.2.7.8.3) combined with any additional LSBs (see
+   Section 4.2.7.8.4), and with the corresponding sign decoded in
+   Section 4.2.7.8.5.  Additionally, let seed be the current
+   pseudorandom seed, which is initialized to the value decoded from
+   Section 4.2.7.7 for the first sample in the current SILK frame, and
+   updated for each subsequent sample according to the procedure below.
+   Finally, let offset_Q23 be the quantization offset from Table 53.
+   Then the following procedure produces the final reconstructed
+   excitation value, e_Q23[i]:
+
+        e_Q23[i] = (e_raw[i] << 8) - sign(e_raw[i])*20 + offset_Q23;
+            seed = (196314165*seed + 907633515) & 0xFFFFFFFF;
+        e_Q23[i] = (seed & 0x80000000) ? -e_Q23[i] : e_Q23[i];
+            seed = (seed + e_raw[i]) & 0xFFFFFFFF;
+
+   When e_raw[i] is zero, sign() returns 0 by the definition in
+   Section 1.1.4, so the factor of 20 does not get added.  The final
+   e_Q23[i] value may require more than 16 bits per sample, but it will
+   not require more than 23, including the sign.
+
+4.2.7.9.  SILK Frame Reconstruction
+
+   The remainder of the reconstruction process for the frame does not
+   need to be bit-exact, as small errors should only introduce
+   proportionally small distortions.  Although the reference
+   implementation only includes a fixed-point version of the remaining
+
+
+
+
+Valin, et al.                Standards Track                   [Page 98]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   steps, this section describes them in terms of a floating-point
+   version for simplicity.  This produces a signal with a nominal range
+   of -1.0 to 1.0.
+
+   silk_decode_core() (decode_core.c) contains the code for the main
+   reconstruction process.  It proceeds subframe-by-subframe, since
+   quantization gains, LTP parameters, and (in 20 ms SILK frames) LPC
+   coefficients can vary from one to the next.
+
+   Let a_Q12[k] be the LPC coefficients for the current subframe.  If
+   this is the first or second subframe of a 20 ms SILK frame and the
+   LSF interpolation factor, w_Q2 (see Section 4.2.7.5.5), is less than
+   4, then these correspond to the final LPC coefficients produced by
+   Section 4.2.7.5.8 from the interpolated LSF coefficients, n1_Q15[k]
+   (computed in Section 4.2.7.5.5).  Otherwise, they correspond to the
+   final LPC coefficients produced from the uninterpolated LSF
+   coefficients for the current frame, n2_Q15[k].
+
+   Also, let n be the number of samples in a subframe (40 for NB, 60 for
+   MB, and 80 for WB), s be the index of the current subframe in this
+   SILK frame (0 or 1 for 10 ms frames, or 0 to 3 for 20 ms frames), and
+   j be the index of the first sample in the residual corresponding to
+   the current subframe.
+
+4.2.7.9.1.  LTP Synthesis
+
+   For unvoiced frames (see Section 4.2.7.3), the LPC residual for i
+   such that j <= i < (j + n) is simply a normalized copy of the
+   excitation signal, i.e.,
+
+                                       e_Q23[i]
+                             res[i] = ---------
+                                       2.0**23
+
+   Voiced SILK frames, on the other hand, pass the excitation through an
+   LTP filter using the parameters decoded in Section 4.2.7.6 to produce
+   an LPC residual.  The LTP filter requires LPC residual values from
+   before the current subframe as input.  However, since the LPC
+   coefficients may have changed, it obtains this residual by
+   "rewhitening" the corresponding output signal using the LPC
+   coefficients from the current subframe.  Let out[i] for i such that
+   (j - pitch_lags[s] - d_LPC - 2) <= i < j be the fully reconstructed
+   output signal from the last (pitch_lags[s] + d_LPC + 2) samples of
+   previous subframes (see Section 4.2.7.9.2), where pitch_lags[s] is
+   the pitch lag for the current subframe from Section 4.2.7.6.1.
+   Additionally, let lpc[i] for i such that (j - s*n - d_LPC) <= i < j
+   be the fully reconstructed output signal from the last (s*n + d_LPC)
+
+
+
+
+Valin, et al.                Standards Track                   [Page 99]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   samples of previous subframes before clamping (see
+   Section 4.2.7.9.2).  During reconstruction of the first subframe for
+   this channel after either
+
+   o  An uncoded regular SILK frame (if this is the side channel), or
+
+   o  A decoder reset (see Section 4.5.2),
+
+   out[i] and lpc[i] are initially cleared to all zeros.  If this is the
+   third or fourth subframe of a 20 ms SILK frame and the LSF
+   interpolation factor, w_Q2 (see Section 4.2.7.5.5), is less than 4,
+   then let out_end be set to (j - (s-2)*n) and let LTP_scale_Q14 be set
+   to 16384.  Otherwise, set out_end to (j - s*n) and set LTP_scale_Q14
+   to the Q14 LTP scaling value from Section 4.2.7.6.3.  Then, for i
+   such that (j - pitch_lags[s] - 2) <= i < out_end, out[i] is
+   rewhitened into an LPC residual, res[i], via
+
+             4.0*LTP_scale_Q14
+    res[i] = ----------------- * clamp(-1.0,
+                gain_Q16[s]
+                                       d_LPC-1
+                                         __              a_Q12[k]
+                                out[i] - \  out[i-k-1] * --------, 1.0)
+                                         /_               4096.0
+                                         k=0
+
+   This requires storage to buffer up to 306 values of out[i] from
+   previous subframes.  This corresponds to WB with a maximum pitch lag
+   of 18 ms * 16 kHz samples, plus 16 samples for d_LPC, plus 2 samples
+   for the width of the LTP filter.  Then, for i such that
+   out_end <= i < j, lpc[i] is rewhitened into an LPC residual, res[i],
+   via
+
+                                        d_LPC-1
+                    65536.0               __              a_Q12[k]
+         res[i] = ----------- * (lpc[i] - \  lpc[i-k-1] * --------)
+                  gain_Q16[s]             /_               4096.0
+                                          k=0
+
+   This requires storage to buffer up to 256 values of lpc[i] from
+   previous subframes (240 from the current SILK frame and 16 from the
+   previous SILK frame).  This corresponds to WB with up to three
+   previous subframes in the current SILK frame, plus 16 samples for
+   d_LPC.  The astute reader will notice that, given the definition of
+   lpc[i] in Section 4.2.7.9.2, the output of this latter equation is
+   merely a scaled version of the values of res[i] from previous
+   subframes.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 100]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Let e_Q23[i] for j <= i < (j + n) be the excitation for the current
+   subframe, and b_Q7[k] for 0 <= k < 5 be the coefficients of the LTP
+   filter taken from the codebook entry in one of Tables 39 through 41
+   corresponding to the index decoded for the current subframe in
+   Section 4.2.7.6.2.  Then for i such that j <= i < (j + n), the LPC
+   residual is
+
+                            4
+                e_Q23[i]   __                                  b_Q7[k]
+      res[i] = --------- + \  res[i - pitch_lags[s] + 2 - k] * -------
+                2.0**23    /_                                   128.0
+                           k=0
+
+4.2.7.9.2.  LPC Synthesis
+
+   LPC synthesis uses the short-term LPC filter to predict the next
+   output coefficient.  For i such that (j - d_LPC) <= i < j, let lpc[i]
+   be the result of LPC synthesis from the last d_LPC samples of the
+   previous subframe or zeros in the first subframe for this channel
+   after either
+
+   o  An uncoded regular SILK frame (if this is the side channel), or
+
+   o  A decoder reset (see Section 4.5.2).
+
+   Then, for i such that j <= i < (j + n), the result of LPC synthesis
+   for the current subframe is
+
+                                        d_LPC-1
+                   gain_Q16[i]            __              a_Q12[k]
+          lpc[i] = ----------- * res[i] + \  lpc[i-k-1] * --------
+                     65536.0              /_               4096.0
+                                          k=0
+
+   The decoder saves the final d_LPC values, i.e., lpc[i] such that
+   (j + n - d_LPC) <= i < (j + n), to feed into the LPC synthesis of the
+   next subframe.  This requires storage for up to 16 values of lpc[i]
+   (for WB frames).
+
+   Then, the signal is clamped into the final nominal range:
+
+                     out[i] = clamp(-1.0, lpc[i], 1.0)
+
+   This clamping occurs entirely after the LPC synthesis filter has run.
+   The decoder saves the unclamped values, lpc[i], to feed into the LPC
+   filter for the next subframe, but saves the clamped values, out[i],
+   for rewhitening in voiced frames.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 101]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.2.8.  Stereo Unmixing
+
+   For stereo streams, after decoding a frame from each channel, the
+   decoder must convert the mid-side (MS) representation into a left-
+   right (LR) representation.  The function silk_stereo_MS_to_LR
+   (stereo_MS_to_LR.c) implements this process.  In it, the decoder
+   predicts the side channel using a) a simple low-passed version of the
+   mid channel, and b) the unfiltered mid channel, using the prediction
+   weights decoded in Section 4.2.7.1.  This simple low-pass filter
+   imposes a one-sample delay, and the unfiltered mid channel is also
+   delayed by one sample.  In order to allow seamless switching between
+   stereo and mono, mono streams must also impose the same one-sample
+   delay.  The encoder requires an additional one-sample delay for both
+   mono and stereo streams, though an encoder may omit the delay for
+   mono if it knows it will never switch to stereo.
+
+   The unmixing process operates in two phases.  The first phase lasts
+   for 8 ms, during which it interpolates the prediction weights from
+   the previous frame, prev_w0_Q13 and prev_w1_Q13, to the values for
+   the current frame, w0_Q13 and w1_Q13.  The second phase simply uses
+   these weights for the remainder of the frame.
+
+   Let mid[i] and side[i] be the contents of out[i] (from
+   Section 4.2.7.9.2) for the current mid and side channels,
+   respectively, and let left[i] and right[i] be the corresponding
+   stereo output channels.  If the side channel is not coded (see
+   Section 4.2.7.2), then side[i] is set to zero.  Also, let j be
+   defined as in Section 4.2.7.9, n1 be the number of samples in phase 1
+   (64 for NB, 96 for MB, and 128 for WB), and n2 be the total number of
+   samples in the frame.  Then, for i such that j <= i < (j + n2), the
+   left and right channel output is
+
+                   prev_w0_Q13                  (w0_Q13 - prev_w0_Q13)
+             w0 =  ----------- + min(i - j, n1)*----------------------
+                     8192.0                           8192.0*n1
+
+                   prev_w1_Q13                  (w1_Q13 - prev_w1_Q13)
+             w1 =  ----------- + min(i - j, n1)*----------------------
+                     8192.0                            8192.0*n1
+
+                  mid[i-2] + 2*mid[i-1] + mid[i]
+             p0 = ------------------------------
+                               4.0
+
+      left[i] = clamp(-1.0, (1 + w1)*mid[i-1] + side[i-1] + w0*p0, 1.0)
+
+     right[i] = clamp(-1.0, (1 - w1)*mid[i-1] - side[i-1] - w0*p0, 1.0)
+
+
+
+
+Valin, et al.                Standards Track                  [Page 102]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   These formulas require two samples prior to index j, the start of the
+   frame, for the mid channel, and one prior sample for the side
+   channel.  For the first frame after a decoder reset, zeros are used
+   instead.
+
+4.2.9.  Resampling
+
+   After stereo unmixing (if any), the decoder applies resampling to
+   convert the decoded SILK output to the sample rate desired by the
+   application.  This is necessary when decoding a Hybrid frame at SWB
+   or FB sample rates, or whenever the decoder wants the output at a
+   different sample rate than the internal SILK sampling rate (e.g., to
+   allow a constant sample rate when the audio bandwidth changes, or to
+   allow mixing with audio from other applications).  The resampler
+   itself is non-normative, and a decoder can use any method it wants to
+   perform the resampling.
+
+   However, a minimum amount of delay is imposed to allow the resampler
+   to operate, and this delay is normative, so that the corresponding
+   delay can be applied to the MDCT layer in the encoder.  A decoder is
+   always free to use a resampler that requires more delay than allowed
+   for here (e.g., to improve quality), but it must then delay the
+   output of the MDCT layer by this extra amount.  Keeping as much delay
+   as possible on the encoder side allows an encoder that knows it will
+   never use any of the SILK or Hybrid modes to skip this delay.  By
+   contrast, if it were all applied by the decoder, then a decoder that
+   processes audio in fixed-size blocks would be forced to delay the
+   output of CELT frames just in case of a later switch to a SILK or
+   Hybrid mode.
+
+   Table 54 gives the maximum resampler delay in samples at 48 kHz for
+   each SILK audio bandwidth.  Because the actual output rate may not be
+   48 kHz, it may not be possible to achieve exactly these delays while
+   using a whole number of input or output samples.  The reference
+   implementation is able to resample to any of the supported output
+   sampling rates (8, 12, 16, 24, or 48 kHz) within or near this delay
+   constraint.  Some resampling filters (including those used by the
+   reference implementation) may add a delay that is not an exact
+   integer, or is not linear-phase, and so cannot be represented by a
+   single delay at all frequencies.  However, such deviations are
+   unlikely to be perceptible, and the comparison tool described in
+   Section 6 is designed to be relatively insensitive to them.  The
+   delays listed here are the ones that should be targeted by the
+   encoder.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 103]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                +-----------------+-----------------------+
+                | Audio Bandwidth | Delay in Milliseconds |
+                +-----------------+-----------------------+
+                | NB              | 0.538                 |
+                |                 |                       |
+                | MB              | 0.692                 |
+                |                 |                       |
+                | WB              | 0.706                 |
+                +-----------------+-----------------------+
+
+                Table 54: SILK Resampler Delay Allocations
+
+   NB is given a smaller decoder delay allocation than MB and WB to
+   allow a higher-order filter when resampling to 8 kHz in both the
+   encoder and decoder.  This implies that the audio content of two SILK
+   frames operating at different bandwidths is not perfectly aligned in
+   time.  This is not an issue for any transitions described in
+   Section 4.5, because they all involve a SILK decoder reset.  When the
+   decoder is reset, any samples remaining in the resampling buffer are
+   discarded, and the resampler is re-initialized with silence.
+
+4.3.  CELT Decoder
+
+   The CELT layer of Opus is based on the Modified Discrete Cosine
+   Transform [MDCT] with partially overlapping windows of 5 to 22.5 ms.
+   The main principle behind CELT is that the MDCT spectrum is divided
+   into bands that (roughly) follow the Bark scale, i.e., the scale of
+   the ear's critical bands [ZWICKER61].  The normal CELT layer uses 21
+   of those bands, though Opus Custom (see Section 6.2) may use a
+   different number of bands.  In Hybrid mode, the first 17 bands (up to
+   8 kHz) are not coded.  A band can contain as little as one MDCT bin
+   per channel, and as many as 176 bins per channel, as detailed in
+   Table 55.  In each band, the gain (energy) is coded separately from
+   the shape of the spectrum.  Coding the gain explicitly makes it easy
+   to preserve the spectral envelope of the signal.  The remaining unit-
+   norm shape vector is encoded using a Pyramid Vector Quantizer
+   (PVQ) Section 4.3.4.
+
+   +--------+--------+------+-------+-------+-------------+------------+
+   | Frame  | 2.5 ms | 5 ms | 10 ms | 20 ms |       Start |       Stop |
+   | Size:  |        |      |       |       |   Frequency |  Frequency |
+   +--------+--------+------+-------+-------+-------------+------------+
+   | Band   |  Bins: |      |       |       |             |            |
+   |        |        |      |       |       |             |            |
+   | 0      |      1 |    2 |     4 |     8 |        0 Hz |     200 Hz |
+   |        |        |      |       |       |             |            |
+   | 1      |      1 |    2 |     4 |     8 |      200 Hz |     400 Hz |
+   |        |        |      |       |       |             |            |
+
+
+
+Valin, et al.                Standards Track                  [Page 104]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   | 2      |      1 |    2 |     4 |     8 |      400 Hz |     600 Hz |
+   |        |        |      |       |       |             |            |
+   | 3      |      1 |    2 |     4 |     8 |      600 Hz |     800 Hz |
+   |        |        |      |       |       |             |            |
+   | 4      |      1 |    2 |     4 |     8 |      800 Hz |    1000 Hz |
+   |        |        |      |       |       |             |            |
+   | 5      |      1 |    2 |     4 |     8 |     1000 Hz |    1200 Hz |
+   |        |        |      |       |       |             |            |
+   | 6      |      1 |    2 |     4 |     8 |     1200 Hz |    1400 Hz |
+   |        |        |      |       |       |             |            |
+   | 7      |      1 |    2 |     4 |     8 |     1400 Hz |    1600 Hz |
+   |        |        |      |       |       |             |            |
+   | 8      |      2 |    4 |     8 |    16 |     1600 Hz |    2000 Hz |
+   |        |        |      |       |       |             |            |
+   | 9      |      2 |    4 |     8 |    16 |     2000 Hz |    2400 Hz |
+   |        |        |      |       |       |             |            |
+   | 10     |      2 |    4 |     8 |    16 |     2400 Hz |    2800 Hz |
+   |        |        |      |       |       |             |            |
+   | 11     |      2 |    4 |     8 |    16 |     2800 Hz |    3200 Hz |
+   |        |        |      |       |       |             |            |
+   | 12     |      4 |    8 |    16 |    32 |     3200 Hz |    4000 Hz |
+   |        |        |      |       |       |             |            |
+   | 13     |      4 |    8 |    16 |    32 |     4000 Hz |    4800 Hz |
+   |        |        |      |       |       |             |            |
+   | 14     |      4 |    8 |    16 |    32 |     4800 Hz |    5600 Hz |
+   |        |        |      |       |       |             |            |
+   | 15     |      6 |   12 |    24 |    48 |     5600 Hz |    6800 Hz |
+   |        |        |      |       |       |             |            |
+   | 16     |      6 |   12 |    24 |    48 |     6800 Hz |    8000 Hz |
+   |        |        |      |       |       |             |            |
+   | 17     |      8 |   16 |    32 |    64 |     8000 Hz |    9600 Hz |
+   |        |        |      |       |       |             |            |
+   | 18     |     12 |   24 |    48 |    96 |     9600 Hz |   12000 Hz |
+   |        |        |      |       |       |             |            |
+   | 19     |     18 |   36 |    72 |   144 |    12000 Hz |   15600 Hz |
+   |        |        |      |       |       |             |            |
+   | 20     |     22 |   44 |    88 |   176 |    15600 Hz |   20000 Hz |
+   +--------+--------+------+-------+-------+-------------+------------+
+
+       Table 55: MDCT Bins per Channel per Band for Each Frame Size
+
+   Transients are notoriously difficult for transform codecs to code.
+   CELT uses two different strategies for them:
+
+   1.  Using multiple smaller MDCTs instead of a single large MDCT, and
+
+   2.  Dynamic time-frequency resolution changes (See Section 4.3.4.5).
+
+
+
+
+Valin, et al.                Standards Track                  [Page 105]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   To improve quality on highly tonal and periodic signals, CELT
+   includes a pre-filter/post-filter combination.  The pre-filter on the
+   encoder side attenuates the signal's harmonics.  The post-filter on
+   the decoder side restores the original gain of the harmonics, while
+   shaping the coding noise to roughly follow the harmonics.  Such noise
+   shaping reduces the perception of the noise.
+
+   When coding a stereo signal, three coding methods are available:
+
+   o  mid-side stereo: encodes the mean and the difference of the left
+      and right channels,
+
+   o  intensity stereo: only encodes the mean of the left and right
+      channels (discards the difference),
+
+   o  dual stereo: encodes the left and right channels separately.
+
+   An overview of the decoder is given in Figure 17.
+
+                       +---------+
+                       | Coarse  |
+                    +->| decoder |----+
+                    |  +---------+    |
+                    |                 |
+                    |  +---------+    v
+                    |  |  Fine   |  +---+
+                    +->| decoder |->| + |
+                    |  +---------+  +---+
+                    |       ^         |
+        +---------+ |       |         |
+        |  Range  | | +----------+    v
+        | Decoder |-+ |   Bit    | +------+
+        +---------+ | |Allocation| | 2**x |
+                    | +----------+ +------+
+                    |       |         |
+                    |       v         v               +--------+
+                    |  +---------+  +---+  +-------+  | pitch  |
+                    +->|   PVQ   |->| * |->| IMDCT |->| post-  |--->
+                    |  | decoder |  +---+  +-------+  | filter |
+                    |  +---------+                    +--------+
+                    |                                      ^
+                    +--------------------------------------+
+
+        Legend: IMDCT = Inverse MDCT
+
+                 Figure 17: Structure of the CELT decoder
+
+   The decoder is based on the following symbols and sets of symbols:
+
+
+
+Valin, et al.                Standards Track                  [Page 106]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+          +---------------+---------------------+---------------+
+          |   Symbol(s)   |         PDF         |   Condition   |
+          +---------------+---------------------+---------------+
+          |    silence    |   {32767, 1}/32768  |               |
+          |               |                     |               |
+          |  post-filter  |       {1, 1}/2      |               |
+          |               |                     |               |
+          |     octave    |     uniform (6)     |  post-filter  |
+          |               |                     |               |
+          |     period    | raw bits (4+octave) |  post-filter  |
+          |               |                     |               |
+          |      gain     |     raw bits (3)    |  post-filter  |
+          |               |                     |               |
+          |     tapset    |     {2, 1, 1}/4     |  post-filter  |
+          |               |                     |               |
+          |   transient   |       {7, 1}/8      |               |
+          |               |                     |               |
+          |     intra     |       {7, 1}/8      |               |
+          |               |                     |               |
+          | coarse energy |    Section 4.3.2    |               |
+          |               |                     |               |
+          |   tf_change   |    Section 4.3.1    |               |
+          |               |                     |               |
+          |   tf_select   |       {1, 1}/2      | Section 4.3.1 |
+          |               |                     |               |
+          |     spread    |   {7, 2, 21, 2}/32  |               |
+          |               |                     |               |
+          |  dyn. alloc.  |    Section 4.3.3    |               |
+          |               |                     |               |
+          |  alloc. trim  |       Table 58      |               |
+          |               |                     |               |
+          |      skip     |       {1, 1}/2      | Section 4.3.3 |
+          |               |                     |               |
+          |   intensity   |       uniform       | Section 4.3.3 |
+          |               |                     |               |
+          |      dual     |       {1, 1}/2      |               |
+          |               |                     |               |
+          |  fine energy  |    Section 4.3.2    |               |
+          |               |                     |               |
+          |    residual   |    Section 4.3.4    |               |
+          |               |                     |               |
+          | anti-collapse |       {1, 1}/2      | Section 4.3.5 |
+          |               |                     |               |
+          |    finalize   |    Section 4.3.2    |               |
+          +---------------+---------------------+---------------+
+
+    Table 56: Order of the Symbols in the CELT Section of the Bitstream
+
+
+
+
+Valin, et al.                Standards Track                  [Page 107]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The decoder extracts information from the range-coded bitstream in
+   the order described in Table 56.  In some circumstances, it is
+   possible for a decoded value to be out of range due to a very small
+   amount of redundancy in the encoding of large integers by the range
+   coder.  In that case, the decoder should assume there has been an
+   error in the coding, decoding, or transmission and SHOULD take
+   measures to conceal the error and/or report to the application that a
+   problem has occurred.  Such out of range errors cannot occur in the
+   SILK layer.
+
+4.3.1.  Transient Decoding
+
+   The "transient" flag indicates whether the frame uses a single long
+   MDCT or several short MDCTs.  When it is set, then the MDCT
+   coefficients represent multiple short MDCTs in the frame.  When not
+   set, the coefficients represent a single long MDCT for the frame.
+   The flag is encoded in the bitstream with a probability of 1/8.  In
+   addition to the global transient flag is a per-band binary flag to
+   change the time-frequency (tf) resolution independently in each band.
+   The change in tf resolution is defined in tf_select_table[][] in
+   celt.c and depends on the frame size, whether the transient flag is
+   set, and the value of tf_select.  The tf_select flag uses a 1/2
+   probability, but is only decoded if it can have an impact on the
+   result knowing the value of all per-band tf_change flags.
+
+4.3.2.  Energy Envelope Decoding
+
+   It is important to quantize the energy with sufficient resolution
+   because any energy quantization error cannot be compensated for at a
+   later stage.  Regardless of the resolution used for encoding the
+   spectral shape of a band, it is perceptually important to preserve
+   the energy in each band.  CELT uses a three-step coarse-fine-fine
+   strategy for encoding the energy in the base-2 log domain, as
+   implemented in quant_bands.c.
+
+4.3.2.1.  Coarse Energy Decoding
+
+   Coarse quantization of the energy uses a fixed resolution of 6 dB
+   (integer part of base-2 log).  To minimize the bitrate, prediction is
+   applied both in time (using the previous frame) and in frequency
+   (using the previous bands).  The part of the prediction that is based
+   on the previous frame can be disabled, creating an "intra" frame
+   where the energy is coded without reference to prior frames.  The
+   decoder first reads the intra flag to determine what prediction is
+   used.  The 2-D z-transform [Z-TRANSFORM] of the prediction filter is
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 108]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                                            -1          -1
+                              (1 - alpha*z_l  )*(1 - z_b  )
+                A(z_l, z_b) = -----------------------------
+                                                 -1
+                                     1 - beta*z_b
+
+   where b is the band index and l is the frame index.  The prediction
+   coefficients applied depend on the frame size in use when not using
+   intra energy and are alpha=0, beta=4915/32768 when using intra
+   energy.  The time-domain prediction is based on the final fine
+   quantization of the previous frame, while the frequency domain
+   (within the current frame) prediction is based on coarse quantization
+   only (because the fine quantization has not been computed yet).  The
+   prediction is clamped internally so that fixed-point implementations
+   with limited dynamic range always remain in the same state as
+   floating point implementations.  We approximate the ideal probability
+   distribution of the prediction error using a Laplace distribution
+   with separate parameters for each frame size in intra- and inter-
+   frame modes.  These parameters are held in the e_prob_model table in
+   quant_bands.c.  The coarse energy decoding is performed by
+   unquant_coarse_energy() (quant_bands.c).  The decoding of the
+   Laplace-distributed values is implemented in ec_laplace_decode()
+   (laplace.c).
+
+4.3.2.2.  Fine Energy Quantization
+
+   The number of bits assigned to fine energy quantization in each band
+   is determined by the bit allocation computation described in
+   Section 4.3.3.  Let B_i be the number of fine energy bits for band i;
+   the refinement is an integer f in the range [0,2**B_i-1].  The
+   mapping between f and the correction applied to the coarse energy is
+   equal to (f+1/2)/2**B_i - 1/2.  Fine energy quantization is
+   implemented in quant_fine_energy() (quant_bands.c).
+
+   When some bits are left "unused" after all other flags have been
+   decoded, these bits are assigned to a "final" step of fine
+   allocation.  In effect, these bits are used to add one extra fine
+   energy bit per band per channel.  The allocation process determines
+   two "priorities" for the final fine bits.  Any remaining bits are
+   first assigned only to bands of priority 0, starting from band 0 and
+   going up.  If all bands of priority 0 have received one bit per
+   channel, then bands of priority 1 are assigned an extra bit per
+   channel, starting from band 0.  If any bits are left after this, they
+   are left unused.  This is implemented in unquant_energy_finalise()
+   (quant_bands.c).
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 109]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.3.3.  Bit Allocation
+
+   Because the bit allocation drives the decoding of the range-coder
+   stream, it MUST be recovered exactly so that identical coding
+   decisions are made in the encoder and decoder.  Any deviation from
+   the reference's resulting bit allocation will result in corrupted
+   output, though implementers are free to implement the procedure in
+   any way that produces identical results.
+
+   The per-band gain-shape structure of the CELT layer ensures that
+   using the same number of bits for the spectral shape of a band in
+   every frame will result in a roughly constant signal-to-noise ratio
+   in that band.  This results in coding noise that has the same
+   spectral envelope as the signal.  The masking curve produced by a
+   standard psychoacoustic model also closely follows the spectral
+   envelope of the signal.  This structure means that the ideal
+   allocation is more consistent from frame to frame than it is for
+   other codecs without an equivalent structure and that a fixed
+   allocation provides fairly consistent perceptual
+   performance [VALIN2010].
+
+   Many codecs transmit significant amounts of side information to
+   control the bit allocation within a frame.  Often this control is
+   only indirect, and it must be exercised carefully to achieve the
+   desired rate constraints.  The CELT layer, however, can adapt over a
+   very wide range of rates, so it has a large number of codebook sizes
+   to choose from for each band.  Explicitly signaling the size of each
+   of these codebooks would impose considerable overhead, even though
+   the allocation is relatively static from frame to frame.  This is
+   because all of the information required to compute these codebook
+   sizes must be derived from a single frame by itself, in order to
+   retain robustness to packet loss, so the signaling cannot take
+   advantage of knowledge of the allocation in neighboring frames.  This
+   problem is exacerbated in low-latency (small frame size)
+   applications, which would include this overhead in every frame.
+
+   For this reason, in the MDCT mode, Opus uses a primarily implicit bit
+   allocation.  The available bitstream capacity is known in advance to
+   both the encoder and decoder without additional signaling, ultimately
+   from the packet sizes expressed by a higher-level protocol.  Using
+   this information, the codec interpolates an allocation from a hard-
+   coded table.
+
+   While the band-energy structure effectively models intra-band
+   masking, it ignores the weaker inter-band masking, band-temporal
+   masking, and other less significant perceptual effects.  While these
+   effects can often be ignored, they can become significant for
+   particular samples.  One mechanism available to encoders would be to
+
+
+
+Valin, et al.                Standards Track                  [Page 110]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   simply increase the overall rate for these frames, but this is not
+   possible in a constant rate mode and can be fairly inefficient.  As a
+   result three explicitly signaled mechanisms are provided to alter the
+   implicit allocation:
+
+   o  Band boost
+
+   o  Allocation trim
+
+   o  Band skipping
+
+   The first of these mechanisms, band boost, allows an encoder to boost
+   the allocation in specific bands.  The second, allocation trim, works
+   by biasing the overall allocation towards higher or lower frequency
+   bands.  The third, band skipping, selects which low-precision high
+   frequency bands will be allocated no shape bits at all.
+
+   In stereo mode, there are two additional parameters potentially coded
+   as part of the allocation procedure: a parameter to allow the
+   selective elimination of allocation for the 'side' (i.e., intensity
+   stereo) in jointly coded bands, and a flag to deactivate joint coding
+   (i.e., dual stereo).  These values are not signaled if they would be
+   meaningless in the overall context of the allocation.
+
+   Because every signaled adjustment increases overhead and
+   implementation complexity, none were included speculatively: the
+   reference encoder makes use of all of these mechanisms.  While the
+   decision logic in the reference was found to be effective enough to
+   justify the overhead and complexity, further analysis techniques may
+   be discovered that increase the effectiveness of these parameters.
+   As with other signaled parameters, an encoder is free to choose the
+   values in any manner, but, unless a technique is known to deliver
+   superior perceptual results, the methods used by the reference
+   implementation should be used.
+
+   The allocation process consists of the following steps: determining
+   the per-band maximum allocation vector, decoding the boosts, decoding
+   the tilt, determining the remaining capacity of the frame, searching
+   the mode table for the entry nearest but not exceeding the available
+   space (subject to the tilt, boosts, band maximums, and band
+   minimums), linear interpolation, reallocation of unused bits with
+   concurrent skip decoding, determination of the fine-energy vs. shape
+   split, and final reallocation.  This process results in a per-band
+   shape allocation (in 1/8th-bit units), a per-band fine-energy
+   allocation (in 1 bit per channel units), a set of band priorities for
+   controlling the use of remaining bits at the end of the frame, and a
+   remaining balance of unallocated space, which is usually zero except
+   at very high rates.
+
+
+
+Valin, et al.                Standards Track                  [Page 111]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The "static" bit allocation (in 1/8 bits) for a quality q, excluding
+   the minimums, maximums, tilt and boosts, is equal to
+   channels*N*alloc[band][q]<<LM>>2, where alloc[][] is given in
+   Table 57 and LM=log2(frame_size/120).  The allocation is obtained by
+   linearly interpolating between two values of q (in steps of 1/64) to
+   find the highest allocation that does not exceed the number of bits
+   remaining.
+
+    Rows indicate the MDCT bands, columns are the different quality (q)
+             parameters.  The units are 1/32 bit per MDCT bin.
+
+     +---+----+-----+-----+-----+-----+-----+-----+-----+-----+-----+
+     | 0 |  1 |   2 |   3 |   4 |   5 |   6 |   7 |   8 |   9 |  10 |
+     +---+----+-----+-----+-----+-----+-----+-----+-----+-----+-----+
+     | 0 | 90 | 110 | 118 | 126 | 134 | 144 | 152 | 162 | 172 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 80 | 100 | 110 | 119 | 127 | 137 | 145 | 155 | 165 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 75 |  90 | 103 | 112 | 120 | 130 | 138 | 148 | 158 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 69 |  84 |  93 | 104 | 114 | 124 | 132 | 142 | 152 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 63 |  78 |  86 |  95 | 103 | 113 | 123 | 133 | 143 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 56 |  71 |  80 |  89 |  97 | 107 | 117 | 127 | 137 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 49 |  65 |  75 |  83 |  91 | 101 | 111 | 121 | 131 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 40 |  58 |  70 |  78 |  85 |  95 | 105 | 115 | 125 | 200 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 34 |  51 |  65 |  72 |  78 |  88 |  98 | 108 | 118 | 198 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 29 |  45 |  59 |  66 |  72 |  82 |  92 | 102 | 112 | 193 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 20 |  39 |  53 |  60 |  66 |  76 |  86 |  96 | 106 | 188 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 18 |  32 |  47 |  54 |  60 |  70 |  80 |  90 | 100 | 183 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 | 10 |  26 |  40 |  47 |  54 |  64 |  74 |  84 |  94 | 178 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |  20 |  31 |  39 |  47 |  57 |  67 |  77 |  87 | 173 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |  12 |  23 |  32 |  41 |  51 |  61 |  71 |  81 | 168 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |   0 |  15 |  25 |  35 |  45 |  55 |  65 |  75 | 163 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |   0 |   4 |  17 |  29 |  39 |  49 |  59 |  69 | 158 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+
+
+
+Valin, et al.                Standards Track                  [Page 112]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+     | 0 |  0 |   0 |   0 |  12 |  23 |  33 |  43 |  53 |  63 | 153 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |   0 |   0 |   1 |  16 |  26 |  36 |  46 |  56 | 148 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |   0 |   0 |   0 |  10 |  15 |  20 |  30 |  45 | 129 |
+     |   |    |     |     |     |     |     |     |     |     |     |
+     | 0 |  0 |   0 |   0 |   0 |   1 |   1 |   1 |   1 |  20 | 104 |
+     +---+----+-----+-----+-----+-----+-----+-----+-----+-----+-----+
+
+                  Table 57: CELT Static Allocation Table
+
+   The maximum allocation vector is an approximation of the maximum
+   space that can be used by each band for a given mode.  The value is
+   approximate because the shape encoding is variable rate (due to
+   entropy coding of splitting parameters).  Setting the maximum too low
+   reduces the maximum achievable quality in a band while setting it too
+   high may result in waste: bitstream capacity available at the end of
+   the frame that cannot be put to any use.  The maximums specified by
+   the codec reflect the average maximum.  In the reference
+   implementation, the maximums in bits/sample are precomputed in a
+   static table (see cache_caps50[] in static_modes_float.h) for each
+   band, for each value of LM, and for both mono and stereo.
+   Implementations are expected to simply use the same table data, but
+   the procedure for generating this table is included in rate.c as part
+   of compute_pulse_cache().
+
+   To convert the values in cache.caps into the actual maximums: first,
+   set nbBands to the maximum number of bands for this mode, and stereo
+   to zero if stereo is not in use and one otherwise.  For each band,
+   set N to the number of MDCT bins covered by the band (for one
+   channel), set LM to the shift value for the frame size.  Then, set i
+   to nbBands*(2*LM+stereo).  Next, set the maximum for the band to the
+   i-th index of cache.caps + 64 and multiply by the number of channels
+   in the current frame (one or two) and by N, then divide the result by
+   4 using integer division.  The resulting vector will be called cap[].
+   The elements fit in signed 16-bit integers but do not fit in 8 bits.
+   This procedure is implemented in the reference in the function
+   init_caps() in celt.c.
+
+   The band boosts are represented by a series of binary symbols that
+   are entropy coded with very low probability.  Each band can
+   potentially be boosted multiple times, subject to the frame actually
+   having enough room to obey the boost and having enough room to code
+   the boost symbol.  The default coding cost for a boost starts out at
+   six bits (probability p=1/64), but subsequent boosts in a band cost
+   only a single bit and every time a band is boosted the initial cost
+   is reduced (down to a minimum of two bits, or p=1/4).  Since the
+
+
+
+
+Valin, et al.                Standards Track                  [Page 113]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   initial cost of coding a boost is 6 bits, the coding cost of the
+   boost symbols when completely unused is 0.48 bits/frame for a 21 band
+   mode (21*-log2(1-1/2**6)).
+
+   To decode the band boosts: First, set 'dynalloc_logp' to 6, the
+   initial amount of storage required to signal a boost in bits,
+   'total_bits' to the size of the frame in 8th bits, 'total_boost' to
+   zero, and 'tell' to the total number of 8th bits decoded so far.  For
+   each band from the coding start (0 normally, but 17 in Hybrid mode)
+   to the coding end (which changes depending on the signaled
+   bandwidth), the boost quanta in units of 1/8 bit is calculated as
+   quanta = min(8*N, max(48, N)).  This represents a boost step size of
+   six bits, subject to a lower limit of 1/8th bit/sample and an upper
+   limit of 1 bit/sample.  Set 'boost' to zero and 'dynalloc_loop_logp'
+   to dynalloc_logp.  While dynalloc_loop_log (the current worst case
+   symbol cost) in 8th bits plus tell is less than total_bits plus
+   total_boost and boost is less than cap[] for this band: Decode a bit
+   from the bitstream with dynalloc_loop_logp as the cost of a one and
+   update tell to reflect the current used capacity.  If the decoded
+   value is zero break the loop.  Otherwise, add quanta to boost and
+   total_boost, subtract quanta from total_bits, and set
+   dynalloc_loop_log to 1.  When the loop finishes 'boost' contains the
+   bit allocation boost for this band.  If boost is non-zero and
+   dynalloc_logp is greater than 2, decrease dynalloc_logp.  Once this
+   process has been executed on all bands, the band boosts have been
+   decoded.  This procedure is implemented around line 2474 of celt.c.
+
+   At very low rates, it is possible that there won't be enough
+   available space to execute the inner loop even once.  In these cases,
+   band boost is not possible, but its overhead is completely
+   eliminated.  Because of the high cost of band boost when activated, a
+   reasonable encoder should not be using it at very low rates.  The
+   reference implements its dynalloc decision logic around line 1304 of
+   celt.c.
+
+   The allocation trim is an integer value from 0-10.  The default value
+   of 5 indicates no trim.  The trim parameter is entropy coded in order
+   to lower the coding cost of less extreme adjustments.  Values lower
+   than 5 bias the allocation towards lower frequencies and values above
+   5 bias it towards higher frequencies.  Like other signaled
+   parameters, signaling of the trim is gated so that it is not included
+   if there is insufficient space available in the bitstream.  To decode
+   the trim, first set the trim value to 5, then if and only if the
+   count of decoded 8th bits so far (ec_tell_frac) plus 48 (6 bits) is
+   less than or equal to the total frame size in 8th bits minus
+   total_boost (a product of the above band boost procedure), decode the
+   trim value using the PDF in Table 58.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 114]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+              +--------------------------------------------+
+              | PDF                                        |
+              +--------------------------------------------+
+              | {2, 2, 5, 10, 22, 46, 22, 10, 5, 2, 2}/128 |
+              +--------------------------------------------+
+
+                        Table 58: PDF for the Trim
+
+   For 10 ms and 20 ms frames using short blocks and that have at least
+   LM+2 bits left prior to the allocation process, one anti-collapse bit
+   is reserved in the allocation process so it can be decoded later.
+   Following the anti-collapse reservation, one bit is reserved for skip
+   if available.
+
+   For stereo frames, bits are reserved for intensity stereo and for
+   dual stereo.  Intensity stereo requires ilog2(end-start) bits.  Those
+   bits are reserved if there are enough bits left.  Following this, one
+   bit is reserved for dual stereo if available.
+
+   The allocation computation begins by setting up some initial
+   conditions. 'total' is set to the remaining available 8th bits,
+   computed by taking the size of the coded frame times 8 and
+   subtracting ec_tell_frac().  From this value, one (8th bit) is
+   subtracted to ensure that the resulting allocation will be
+   conservative. 'anti_collapse_rsv' is set to 8 (8th bits) if and only
+   if the frame is a transient, LM is greater than 1, and total is
+   greater than or equal to (LM+2) * 8.  Total is then decremented by
+   anti_collapse_rsv and clamped to be equal to or greater than zero.
+   'skip_rsv' is set to 8 (8th bits) if total is greater than 8,
+   otherwise it is zero.  Total is then decremented by skip_rsv.  This
+   reserves space for the final skipping flag.
+
+   If the current frame is stereo, intensity_rsv is set to the
+   conservative log2 in 8th bits of the number of coded bands for this
+   frame (given by the table LOG2_FRAC_TABLE in rate.c).  If
+   intensity_rsv is greater than total, then intensity_rsv is set to
+   zero.  Otherwise, total is decremented by intensity_rsv, and if total
+   is still greater than 8, dual_stereo_rsv is set to 8 and total is
+   decremented by dual_stereo_rsv.
+
+   The allocation process then computes a vector representing the hard
+   minimum amounts allocation any band will receive for shape.  This
+   minimum is higher than the technical limit of the PVQ process, but
+   very low rate allocations produce an excessively sparse spectrum and
+   these bands are better served by having no allocation at all.  For
+   each coded band, set thresh[band] to 24 times the number of MDCT bins
+   in the band and divide by 16.  If 8 times the number of channels is
+   greater, use that instead.  This sets the minimum allocation to one
+
+
+
+Valin, et al.                Standards Track                  [Page 115]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   bit per channel or 48 128th bits per MDCT bin, whichever is greater.
+   The band-size dependent part of this value is not scaled by the
+   channel count, because at the very low rates where this limit is
+   applicable there will usually be no bits allocated to the side.
+
+   The previously decoded allocation trim is used to derive a vector of
+   per-band adjustments, 'trim_offsets[]'.  For each coded band take the
+   alloc_trim and subtract 5 and LM.  Then, multiply the result by the
+   number of channels, the number of MDCT bins in the shortest frame
+   size for this mode, the number of remaining bands, 2**LM, and 8.
+   Next, divide this value by 64.  Finally, if the number of MDCT bins
+   in the band per channel is only one, 8 times the number of channels
+   is subtracted in order to diminish the allocation by one bit, because
+   width 1 bands receive greater benefit from the coarse energy coding.
+
+4.3.4.  Shape Decoding
+
+   In each band, the normalized "shape" is encoded using Pyramid Vector
+   Quantizer.
+
+   In the simplest case, the number of bits allocated in Section 4.3.3
+   is converted to a number of pulses as described by Section 4.3.4.1.
+   Knowing the number of pulses and the number of samples in the band,
+   the decoder calculates the size of the codebook as detailed in
+   Section 4.3.4.2.  The size is used to decode an unsigned integer
+   (uniform probability model), which is the codeword index.  This index
+   is converted into the corresponding vector as explained in
+   Section 4.3.4.2.  This vector is then scaled to unit norm.
+
+4.3.4.1.  Bits to Pulses
+
+   Although the allocation is performed in 1/8th bit units, the
+   quantization requires an integer number of pulses K.  To do this, the
+   encoder searches for the value of K that produces the number of bits
+   nearest to the allocated value (rounding down if exactly halfway
+   between two values), not to exceed the total number of bits
+   available.  For efficiency reasons, the search is performed against a
+   precomputed allocation table that only permits some K values for each
+   N.  The number of codebook entries can be computed as explained in
+   Section 4.3.4.2.  The difference between the number of bits allocated
+   and the number of bits used is accumulated to a "balance"
+   (initialized to zero) that helps adjust the allocation for the next
+   bands.  One third of the balance is applied to the bit allocation of
+   each band to help achieve the target allocation.  The only exceptions
+   are the band before the last and the last band, for which half the
+   balance and the whole balance are applied, respectively.
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 116]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.3.4.2.  PVQ Decoding
+
+   Decoding of PVQ vectors is implemented in decode_pulses() (cwrs.c).
+   The unique codeword index is decoded as a uniformly distributed
+   integer value between 0 and V(N,K)-1, where V(N,K) is the number of
+   possible combinations of K pulses in N samples.  The index is then
+   converted to a vector in the same way specified in [PVQ].  The
+   indexing is based on the calculation of V(N,K) (denoted N(L,K) in
+   [PVQ]).
+
+   The number of combinations can be computed recursively as V(N,K) =
+   V(N-1,K) + V(N,K-1) + V(N-1,K-1), with V(N,0) = 1 and V(0,K) = 0, K
+   != 0.  There are many different ways to compute V(N,K), including
+   precomputed tables and direct use of the recursive formulation.  The
+   reference implementation applies the recursive formulation one line
+   (or column) at a time to save on memory use, along with an alternate,
+   univariate recurrence to initialize an arbitrary line, and direct
+   polynomial solutions for small N.  All of these methods are
+   equivalent, and have different trade-offs in speed, memory usage, and
+   code size.  Implementations MAY use any methods they like, as long as
+   they are equivalent to the mathematical definition.
+
+   The decoded vector X is recovered as follows.  Let i be the index
+   decoded with the procedure in Section 4.1.5 with ft = V(N,K), so that
+   0 <= i < V(N,K).  Let k = K.  Then, for j = 0 to (N - 1), inclusive,
+   do:
+
+   1.  Let p = (V(N-j-1,k) + V(N-j,k))/2.
+
+   2.  If i < p, then let sgn = 1, else let sgn = -1 and set i = i - p.
+
+   3.  Let k0 = k and set p = p - V(N-j-1,k).
+
+   4.  While p > i, set k = k - 1 and p = p - V(N-j-1,k).
+
+   5.  Set X[j] = sgn*(k0 - k) and i = i - p.
+
+   The decoded vector X is then normalized such that its L2-norm equals
+   one.
+
+4.3.4.3.  Spreading
+
+   The normalized vector decoded in Section 4.3.4.2 is then rotated for
+   the purpose of avoiding tonal artifacts.  The rotation gain is equal
+   to
+
+                           g_r = N / (N + f_r*K)
+
+
+
+
+Valin, et al.                Standards Track                  [Page 117]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   where N is the number of dimensions, K is the number of pulses, and
+   f_r depends on the value of the "spread" parameter in the bitstream.
+
+                 +--------------+------------------------+
+                 | Spread value | f_r                    |
+                 +--------------+------------------------+
+                 | 0            | infinite (no rotation) |
+                 |              |                        |
+                 | 1            | 15                     |
+                 |              |                        |
+                 | 2            | 10                     |
+                 |              |                        |
+                 | 3            | 5                      |
+                 +--------------+------------------------+
+
+                        Table 59: Spreading Values
+
+   The rotation angle is then calculated as
+
+                                              2
+                                     pi *  g_r
+                             theta = ----------
+                                         4
+
+   A 2-D rotation R(i,j) between points x_i and x_j is defined as:
+
+                  x_i' =  cos(theta)*x_i + sin(theta)*x_j
+                  x_j' = -sin(theta)*x_i + cos(theta)*x_j
+
+   An N-D rotation is then achieved by applying a series of 2-D
+   rotations back and forth, in the following order: R(x_1, x_2), R(x_2,
+   x_3), ..., R(x_N-2, X_N-1), R(x_N-1, X_N), R(x_N-2, X_N-1), ...,
+   R(x_1, x_2).
+
+   If the decoded vector represents more than one time block, then this
+   spreading process is applied separately on each time block.  Also, if
+   each block represents 8 samples or more, then another N-D rotation,
+   by (pi/2-theta), is applied _before_ the rotation described above.
+   This extra rotation is applied in an interleaved manner with a stride
+   equal to round(sqrt(N/nb_blocks)), i.e., it is applied independently
+   for each set of sample S_k = {stride*n + k}, n=0..N/stride-1.
+
+4.3.4.4.  Split Decoding
+
+   To avoid the need for multi-precision calculations when decoding PVQ
+   codevectors, the maximum size allowed for codebooks is 32 bits.  When
+   larger codebooks are needed, the vector is instead split in two sub-
+   vectors of size N/2.  A quantized gain parameter with precision
+
+
+
+Valin, et al.                Standards Track                  [Page 118]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   derived from the current allocation is entropy coded to represent the
+   relative gains of each side of the split, and the entire decoding
+   process is recursively applied.  Multiple levels of splitting may be
+   applied up to a limit of LM+1 splits.  The same recursive mechanism
+   is applied for the joint coding of stereo audio.
+
+4.3.4.5.  Time-Frequency Change
+
+   The time-frequency (TF) parameters are used to control the time-
+   frequency resolution trade-off in each coded band.  For each band,
+   there are two possible TF choices.  For the first band coded, the PDF
+   is {3, 1}/4 for frames marked as transient and {15, 1}/16 for the
+   other frames.  For subsequent bands, the TF choice is coded relative
+   to the previous TF choice with probability {15, 1}/16 for transient
+   frames and {31, 1}/32 otherwise.  The mapping between the decoded TF
+   choices and the adjustment in TF resolution is shown in the tables
+   below.
+
+                       +-----------------+---+----+
+                       | Frame size (ms) | 0 |  1 |
+                       +-----------------+---+----+
+                       |       2.5       | 0 | -1 |
+                       |                 |   |    |
+                       |        5        | 0 | -1 |
+                       |                 |   |    |
+                       |        10       | 0 | -2 |
+                       |                 |   |    |
+                       |        20       | 0 | -2 |
+                       +-----------------+---+----+
+
+     Table 60: TF Adjustments for Non-transient Frames and tf_select=0
+
+                       +-----------------+---+----+
+                       | Frame size (ms) | 0 |  1 |
+                       +-----------------+---+----+
+                       |       2.5       | 0 | -1 |
+                       |                 |   |    |
+                       |        5        | 0 | -2 |
+                       |                 |   |    |
+                       |        10       | 0 | -3 |
+                       |                 |   |    |
+                       |        20       | 0 | -3 |
+                       +-----------------+---+----+
+
+     Table 61: TF Adjustments for Non-transient Frames and tf_select=1
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 119]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                       +-----------------+---+----+
+                       | Frame size (ms) | 0 |  1 |
+                       +-----------------+---+----+
+                       |       2.5       | 0 | -1 |
+                       |                 |   |    |
+                       |        5        | 1 |  0 |
+                       |                 |   |    |
+                       |        10       | 2 |  0 |
+                       |                 |   |    |
+                       |        20       | 3 |  0 |
+                       +-----------------+---+----+
+
+       Table 62: TF Adjustments for Transient Frames and tf_select=0
+
+                       +-----------------+---+----+
+                       | Frame size (ms) | 0 |  1 |
+                       +-----------------+---+----+
+                       |       2.5       | 0 | -1 |
+                       |                 |   |    |
+                       |        5        | 1 | -1 |
+                       |                 |   |    |
+                       |        10       | 1 | -1 |
+                       |                 |   |    |
+                       |        20       | 1 | -1 |
+                       +-----------------+---+----+
+
+       Table 63: TF Adjustments for Transient Frames and tf_select=1
+
+   A negative TF adjustment means that the temporal resolution is
+   increased, while a positive TF adjustment means that the frequency
+   resolution is increased.  Changes in TF resolution are implemented
+   using the Hadamard transform [HADAMARD].  To increase the time
+   resolution by N, N "levels" of the Hadamard transform are applied to
+   the decoded vector for each interleaved MDCT vector.  To increase the
+   frequency resolution (assumes a transient frame), then N levels of
+   the Hadamard transform are applied _across_ the interleaved MDCT
+   vector.  In the case of increased time resolution, the decoder uses
+   the "sequency order" because the input vector is sorted in time.
+
+4.3.5.  Anti-collapse Processing
+
+   The anti-collapse feature is designed to avoid the situation where
+   the use of multiple short MDCTs causes the energy in one or more of
+   the MDCTs to be zero for some bands, causing unpleasant artifacts.
+   When the frame has the transient bit set, an anti-collapse bit is
+   decoded.  When anti-collapse is set, the energy in each small MDCT is
+   prevented from collapsing to zero.  For each band of each MDCT where
+   a collapse is detected, a pseudo-random signal is inserted with an
+
+
+
+Valin, et al.                Standards Track                  [Page 120]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   energy corresponding to the minimum energy over the two previous
+   frames.  A renormalization step is then required to ensure that the
+   anti-collapse step did not alter the energy preservation property.
+
+4.3.6.  Denormalization
+
+   Just as each band was normalized in the encoder, the last step of the
+   decoder before the inverse MDCT is to denormalize the bands.  Each
+   decoded normalized band is multiplied by the square root of the
+   decoded energy.  This is done by denormalise_bands() (bands.c).
+
+4.3.7.  Inverse MDCT
+
+   The inverse MDCT implementation has no special characteristics.  The
+   input is N frequency-domain samples and the output is 2*N time-domain
+   samples, while scaling by 1/2.  A "low-overlap" window reduces the
+   algorithmic delay.  It is derived from a basic (full-overlap) 240-
+   sample version of the window used by the Vorbis codec:
+
+                                                         2
+                          /   /pi      /pi   n + 1/2\ \ \
+                   W(n) = |sin|-- * sin|-- * -------| | |
+                          \   \2       \2       L   / / /
+
+   The low-overlap window is created by zero-padding the basic window
+   and inserting ones in the middle, such that the resulting window
+   still satisfies power complementarity [PRINCEN86].  The IMDCT and
+   windowing are performed by mdct_backward (mdct.c).
+
+4.3.7.1.  Post-Filter
+
+   The output of the inverse MDCT (after weighted overlap-add) is sent
+   to the post-filter.  Although the post-filter is applied at the end,
+   the post-filter parameters are encoded at the beginning, just after
+   the silence flag.  The post-filter can be switched on or off using
+   one bit (logp=1).  If the post-filter is enabled, then the octave is
+   decoded as an integer value between 0 and 6 of uniform probability.
+   Once the octave is known, the fine pitch within the octave is decoded
+   using 4+octave raw bits.  The final pitch period is equal to
+   (16<<octave)+fine_pitch-1 so it is bounded between 15 and 1022,
+   inclusively.  Next, the gain is decoded as three raw bits and is
+   equal to G=3*(int_gain+1)/32.  The set of post-filter taps is decoded
+   last, using a pdf equal to {2, 1, 1}/4.  Tapset zero corresponds to
+   the filter coefficients g0 = 0.3066406250, g1 = 0.2170410156, g2 =
+   0.1296386719.  Tapset one corresponds to the filter coefficients g0 =
+   0.4638671875, g1 = 0.2680664062, g2 = 0, and tapset two uses filter
+   coefficients g0 = 0.7998046875, g1 = 0.1000976562, g2 = 0.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 121]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The post-filter response is thus computed as:
+
+
+             y(n) = x(n) + G*(g0*y(n-T) + g1*(y(n-T+1)+y(n-T+1))
+                                        + g2*(y(n-T+2)+y(n-T+2)))
+
+
+   During a transition between different gains, a smooth transition is
+   calculated using the square of the MDCT window.  It is important that
+   values of y(n) be interpolated one at a time such that the past value
+   of y(n) used is interpolated.
+
+4.3.7.2.  De-emphasis
+
+   After the post-filter, the signal is de-emphasized using the inverse
+   of the pre-emphasis filter used in the encoder:
+
+                            1            1
+                           ---- = ---------------
+                           A(z)                -1
+                                  1 - alpha_p*z
+
+   where alpha_p=0.8500061035.
+
+4.4.  Packet Loss Concealment (PLC)
+
+   Packet Loss Concealment (PLC) is an optional decoder-side feature
+   that SHOULD be included when receiving from an unreliable channel.
+   Because PLC is not part of the bitstream, there are many acceptable
+   ways to implement PLC with different complexity/quality trade-offs.
+
+   The PLC in the reference implementation depends on the mode of last
+   packet received.  In CELT mode, the PLC finds a periodicity in the
+   decoded signal and repeats the windowed waveform using the pitch
+   offset.  The windowed waveform is overlapped in such a way as to
+   preserve the time-domain aliasing cancellation with the previous
+   frame and the next frame.  This is implemented in celt_decode_lost()
+   (mdct.c).  In SILK mode, the PLC uses LPC extrapolation from the
+   previous frame, implemented in silk_PLC() (PLC.c).
+
+4.4.1.  Clock Drift Compensation
+
+   Clock drift refers to the gradual desynchronization of two endpoints
+   whose sample clocks run at different frequencies while they are
+   streaming live audio.  Differences in clock frequencies are generally
+   attributable to manufacturing variation in the endpoints' clock
+   hardware.  For long-lived streams, the time difference between sender
+   and receiver can grow without bound.
+
+
+
+Valin, et al.                Standards Track                  [Page 122]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   When the sender's clock runs slower than the receiver's, the effect
+   is similar to packet loss: too few packets are received.  The
+   receiver can distinguish between drift and loss if the transport
+   provides packet timestamps.  A receiver for live streams SHOULD
+   conceal the effects of drift, and it MAY do so by invoking the PLC.
+
+   When the sender's clock runs faster than the receiver's, too many
+   packets will be received.  The receiver MAY respond by skipping any
+   packet (i.e., not submitting the packet for decoding).  This is
+   likely to produce a less severe artifact than if the frame were
+   dropped after decoding.
+
+   A decoder MAY employ a more sophisticated drift compensation method.
+   For example, the NetEQ component [GOOGLE-NETEQ] of the Google WebRTC
+   codebase [GOOGLE-WEBRTC] compensates for drift by adding or removing
+   one period when the signal is highly periodic.  The reference
+   implementation of Opus allows a caller to learn whether the current
+   frame's signal is highly periodic, and if so what the period is,
+   using the OPUS_GET_PITCH() request.
+
+4.5.  Configuration Switching
+
+   Switching between the Opus coding modes, audio bandwidths, and
+   channel counts requires careful consideration to avoid audible
+   glitches.  Switching between any two configurations of the CELT-only
+   mode, any two configurations of the Hybrid mode, or from WB SILK to
+   Hybrid mode does not require any special treatment in the decoder, as
+   the MDCT overlap will smooth the transition.  Switching from Hybrid
+   mode to WB SILK requires adding in the final contents of the CELT
+   overlap buffer to the first SILK-only packet.  This can be done by
+   decoding a 2.5 ms silence frame with the CELT decoder using the
+   channel count of the SILK-only packet (and any choice of audio
+   bandwidth), which will correctly handle the cases when the channel
+   count changes as well.
+
+   When changing the channel count for SILK-only or Hybrid packets, the
+   encoder can avoid glitches by smoothly varying the stereo width of
+   the input signal before or after the transition, and it SHOULD do so.
+   However, other transitions between SILK-only packets or between NB or
+   MB SILK and Hybrid packets may cause glitches, because neither the
+   LSF coefficients nor the LTP, LPC, stereo unmixing, and resampler
+   buffers are available at the new sample rate.  These switches SHOULD
+   be delayed by the encoder until quiet periods or transients, where
+   the inevitable glitches will be less audible.  Additionally, the
+   bitstream MAY include redundant side information ("redundancy"), in
+   the form of additional CELT frames embedded in each of the Opus
+   frames around the transition.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 123]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The other transitions that cannot be easily handled are those where
+   the lower frequencies switch between the SILK LP-based model and the
+   CELT MDCT model.  However, an encoder may not have an opportunity to
+   delay such a switch to a convenient point.  For example, if the
+   content switches from speech to music, and the encoder does not have
+   enough latency in its analysis to detect this in advance, there may
+   be no convenient silence period during which to make the transition
+   for quite some time.  To avoid or reduce glitches during these
+   problematic mode transitions, and between audio bandwidth changes in
+   the SILK-only modes, transitions MAY include redundant side
+   information ("redundancy"), in the form of an additional CELT frame
+   embedded in the Opus frame.
+
+   A transition between coding the lower frequencies with the LP model
+   and the MDCT model or a transition that involves changing the SILK
+   bandwidth is only normatively specified when it includes redundancy.
+   For those without redundancy, it is RECOMMENDED that the decoder use
+   a concealment technique (e.g., make use of a PLC algorithm) to "fill
+   in" the gap or discontinuity caused by the mode transition.
+   Therefore, PLC MUST NOT be applied during any normative transition,
+   i.e., when
+
+   o  A packet includes redundancy for this transition (as described
+      below),
+
+   o  The transition is between any WB SILK packet and any Hybrid
+      packet, or vice versa,
+
+   o  The transition is between any two Hybrid mode packets, or
+
+   o  The transition is between any two CELT mode packets,
+
+   unless there is actual packet loss.
+
+4.5.1.  Transition Side Information (Redundancy)
+
+   Transitions with side information include an extra 5 ms "redundant"
+   CELT frame within the Opus frame.  This frame is designed to fill in
+   the gap or discontinuity in the different layers without requiring
+   the decoder to conceal it.  For transitions from CELT-only to SILK-
+   only or Hybrid, the redundant frame is inserted in the first Opus
+   frame after the transition (i.e., the first SILK-only or Hybrid
+   frame).  For transitions from SILK-only or Hybrid to CELT-only, the
+   redundant frame is inserted in the last Opus frame before the
+   transition (i.e., the last SILK-only or Hybrid frame).
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 124]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.5.1.1.  Redundancy Flag
+
+   The presence of redundancy is signaled in all SILK-only and Hybrid
+   frames, not just those involved in a mode transition.  This allows
+   the frames to be decoded correctly even if an adjacent frame is lost.
+   For SILK-only frames, this signaling is implicit, based on the size
+   of the Opus frame and the number of bits consumed decoding the SILK
+   portion of it.  After decoding the SILK portion of the Opus frame,
+   the decoder uses ec_tell() (see Section 4.1.6.1) to check if there
+   are at least 17 bits remaining.  If so, then the frame contains
+   redundancy.
+
+   For Hybrid frames, this signaling is explicit.  After decoding the
+   SILK portion of the Opus frame, the decoder uses ec_tell() (see
+   Section 4.1.6.1) to ensure there are at least 37 bits remaining.  If
+   so, it reads a symbol with the PDF in Table 64, and if the value is
+   1, then the frame contains redundancy.  Otherwise (if there were
+   fewer than 37 bits left or the value was 0), the frame does not
+   contain redundancy.
+
+                            +----------------+
+                            | PDF            |
+                            +----------------+
+                            | {4095, 1}/4096 |
+                            +----------------+
+
+                       Table 64: Redundancy Flag PDF
+
+4.5.1.2.  Redundancy Position Flag
+
+   Since the current frame is a SILK-only or a Hybrid frame, it must be
+   at least 10 ms.  Therefore, it needs an additional flag to indicate
+   whether the redundant 5 ms CELT frame should be mixed into the
+   beginning of the current frame, or the end.  After determining that a
+   frame contains redundancy, the decoder reads a 1 bit symbol with a
+   uniform PDF (Table 65).
+
+                               +----------+
+                               | PDF      |
+                               +----------+
+                               | {1, 1}/2 |
+                               +----------+
+
+                     Table 65: Redundancy Position PDF
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 125]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   If the value is zero, this is the first frame in the transition, and
+   the redundancy belongs at the end.  If the value is one, this is the
+   second frame in the transition, and the redundancy belongs at the
+   beginning.  There is no way to specify that an Opus frame contains
+   separate redundant CELT frames at both the beginning and the end.
+
+4.5.1.3.  Redundancy Size
+
+   Unlike the CELT portion of a Hybrid frame, the redundant CELT frame
+   does not use the same entropy coder state as the rest of the Opus
+   frame, because this would break the CELT bit allocation mechanism in
+   Hybrid frames.  Thus, a redundant CELT frame always starts and ends
+   on a byte boundary, even in SILK-only frames, where this is not
+   strictly necessary.
+
+   For SILK-only frames, the number of bytes in the redundant CELT frame
+   is simply the number of whole bytes remaining, which must be at least
+   2, due to the space check in Section 4.5.1.1.  For Hybrid frames, the
+   number of bytes is equal to 2, plus a decoded unsigned integer less
+   than 256 (see Section 4.1.5).  This may be more than the number of
+   whole bytes remaining in the Opus frame, in which case the frame is
+   invalid.  However, a decoder is not required to ignore the entire
+   frame, as this may be the result of a bit error that desynchronized
+   the range coder.  There may still be useful data before the error,
+   and a decoder MAY keep any audio decoded so far instead of invoking
+   the PLC, but it is RECOMMENDED that the decoder stop decoding and
+   discard the rest of the current Opus frame.
+
+   It would have been possible to avoid these invalid states in the
+   design of Opus by limiting the range of the explicit length decoded
+   from Hybrid frames by the actual number of whole bytes remaining.
+   However, this would require an encoder to determine the rate
+   allocation for the MDCT layer up front, before it began encoding that
+   layer.  By allowing some invalid sizes, the encoder is able to defer
+   that decision until much later.  When encoding Hybrid frames that do
+   not include redundancy, the encoder must still decide up front if it
+   wishes to use the minimum 37 bits required to trigger encoding of the
+   redundancy flag, but this is a much looser restriction.
+
+   After determining the size of the redundant CELT frame, the decoder
+   reduces the size of the buffer currently in use by the range coder by
+   that amount.  The MDCT layer reads any raw bits from the end of this
+   reduced buffer, and all calculations of the number of bits remaining
+   in the buffer must be done using this new, reduced size, rather than
+   the original size of the Opus frame.
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 126]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.5.1.4.  Decoding the Redundancy
+
+   The redundant frame is decoded like any other CELT-only frame, with
+   the exception that it does not contain a TOC byte.  The frame size is
+   fixed at 5 ms, the channel count is set to that of the current frame,
+   and the audio bandwidth is also set to that of the current frame,
+   with the exception that for MB SILK frames, it is set to WB.
+
+   If the redundancy belongs at the beginning (in a CELT-only to SILK-
+   only or Hybrid transition), the final reconstructed output uses the
+   first 2.5 ms of audio output by the decoder for the redundant frame
+   as is, discarding the corresponding output from the SILK-only or
+   Hybrid portion of the frame.  The remaining 2.5 ms is cross-lapped
+   with the decoded SILK/Hybrid signal using the CELT's power-
+   complementary MDCT window to ensure a smooth transition.
+
+   If the redundancy belongs at the end (in a SILK-only or Hybrid to
+   CELT-only transition), only the second half (2.5 ms) of the audio
+   output by the decoder for the redundant frame is used.  In that case,
+   the second half of the redundant frame is cross-lapped with the end
+   of the SILK/Hybrid signal, again using CELT's power-complementary
+   MDCT window to ensure a smooth transition.
+
+4.5.2.  State Reset
+
+   When a transition occurs, the state of the SILK or the CELT decoder
+   (or both) may need to be reset before decoding a frame in the new
+   mode.  This avoids reusing "out of date" memory, which may not have
+   been updated in some time or may not be in a well-defined state due
+   to, e.g., PLC.  The SILK state is reset before every SILK-only or
+   Hybrid frame where the previous frame was CELT-only.  The CELT state
+   is reset every time the operating mode changes and the new mode is
+   either Hybrid or CELT-only, except when the transition uses
+   redundancy as described above.  When switching from SILK-only or
+   Hybrid to CELT-only with redundancy, the CELT state is reset before
+   decoding the redundant CELT frame embedded in the SILK-only or Hybrid
+   frame, but it is not reset before decoding the following CELT-only
+   frame.  When switching from CELT-only mode to SILK-only or Hybrid
+   mode with redundancy, the CELT decoder is not reset for decoding the
+   redundant CELT frame.
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 127]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+4.5.3.  Summary of Transitions
+
+   Figure 18 illustrates all of the normative transitions involving a
+   mode change, an audio bandwidth change, or both.  Each one uses an S,
+   H, or C to represent an Opus frame in the corresponding mode.  In
+   addition, an R indicates the presence of redundancy in the Opus frame
+   with which it is cross-lapped.  Its location in the first or last
+   5 ms is assumed to correspond to whether it is the frame before or
+   after the transition.  Other uses of redundancy are non-normative.
+   Finally, a c indicates the contents of the CELT overlap buffer after
+   the previously decoded frame (i.e., as extracted by decoding a
+   silence frame).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 128]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+    SILK to SILK with Redundancy:             S -> S -> S
+                                                        &
+                                                       !R -> R
+                                                             &
+                                                            ;S -> S -> S
+
+    NB or MB SILK to Hybrid with Redundancy:  S -> S -> S
+                                                        &
+                                                       !R ->;H -> H -> H
+
+    WB SILK to Hybrid:                        S -> S -> S ->!H -> H -> H
+
+    SILK to CELT with Redundancy:             S -> S -> S
+                                                        &
+                                                       !R -> C -> C -> C
+
+    Hybrid to NB or MB SILK with Redundancy:  H -> H -> H
+                                                        &
+                                                       !R -> R
+                                                             &
+                                                            ;S -> S -> S
+
+    Hybrid to WB SILK:                        H -> H -> H -> c
+                                                          \  +
+                                                           > S -> S -> S
+
+    Hybrid to CELT with Redundancy:           H -> H -> H
+                                                        &
+                                                       !R -> C -> C -> C
+
+    CELT to SILK with Redundancy:             C -> C -> C -> R
+                                                             &
+                                                            ;S -> S -> S
+
+    CELT to Hybrid with Redundancy:           C -> C -> C -> R
+                                                             &
+                                                            |H -> H -> H
+
+    Key:
+    S   SILK-only frame                 ;   SILK decoder reset
+    H   Hybrid frame                    |   CELT and SILK decoder resets
+    C   CELT-only frame                 !   CELT decoder reset
+    c   CELT overlap                    +   Direct mixing
+    R   Redundant CELT frame            &   Windowed cross-lap
+
+                     Figure 18: Normative Transitions
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 129]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The first two and the last two Opus frames in each example are
+   illustrative, i.e., there is no requirement that a stream remain in
+   the same configuration for three consecutive frames before or after a
+   switch.
+
+   The behavior of transitions without redundancy where PLC is allowed
+   is non-normative.  An encoder might still wish to use these
+   transitions if, for example, it doesn't want to add the extra bitrate
+   required for redundancy or if it makes a decision to switch after it
+   has already transmitted the frame that would have had to contain the
+   redundancy.  Figure 19 illustrates the recommended cross-lapping and
+   decoder resets for these transitions.
+
+    SILK to SILK (audio bandwidth change):    S -> S -> S   ;S -> S -> S
+
+    NB or MB SILK to Hybrid:                  S -> S -> S   |H -> H -> H
+
+    SILK to CELT without Redundancy:          S -> S -> S -> P
+                                                             &
+                                                            !C -> C -> C
+
+    Hybrid to NB or MB SILK:                  H -> H -> H -> c
+                                                             +
+                                                            ;S -> S -> S
+
+    Hybrid to CELT without Redundancy:        H -> H -> H -> P
+                                                             &
+                                                            !C -> C -> C
+
+    CELT to SILK without Redundancy:          C -> C -> C -> P
+                                                             &
+                                                            ;S -> S -> S
+
+    CELT to Hybrid without Redundancy:        C -> C -> C -> P
+                                                             &
+                                                            |H -> H -> H
+
+    Key:
+    S   SILK-only frame                 ;   SILK decoder reset
+    H   Hybrid frame                    |   CELT and SILK decoder resets
+    C   CELT-only frame                 !   CELT decoder reset
+    c   CELT overlap                    +   Direct mixing
+    P   Packet Loss Concealment         &   Windowed cross-lap
+
+             Figure 19: Recommended Non-Normative Transitions
+
+   Encoders SHOULD NOT use other transitions, e.g., those that involve
+   redundancy in ways not illustrated in Figure 18.
+
+
+
+Valin, et al.                Standards Track                  [Page 130]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+5.  Opus Encoder
+
+   Just like the decoder, the Opus encoder also normally consists of two
+   main blocks: the SILK encoder and the CELT encoder.  However, unlike
+   the case of the decoder, a valid (though potentially suboptimal) Opus
+   encoder is not required to support all modes and may thus only
+   include a SILK encoder module or a CELT encoder module.  The output
+   bitstream of the Opus encoding contains bits from the SILK and CELT
+   encoders, though these are not separable due to the use of a range
+   coder.  A block diagram of the encoder is illustrated below.
+
+                        +------------+    +---------+
+                        |   Sample   |    |  SILK   |------+
+                     +->|    Rate    |--->| Encoder |      V
+      +-----------+  |  | Conversion |    |         | +---------+
+      | Optional  |  |  +------------+    +---------+ |  Range  |
+    ->| High-pass |--+                                | Encoder |---->
+      |  Filter   |  |  +--------------+  +---------+ |         | Bit-
+      +-----------+  |  |    Delay     |  |  CELT   | +---------+ stream
+                     +->| Compensation |->| Encoder |      ^
+                        |              |  |         |------+
+                        +--------------+  +---------+
+
+                          Figure 20: Opus Encoder
+
+   For a normal encoder where both the SILK and the CELT modules are
+   included, an optimal encoder should select which coding mode to use
+   at run-time depending on the conditions.  In the reference
+   implementation, the frame size is selected by the application, but
+   the other configuration parameters (number of channels, bandwidth,
+   mode) are automatically selected (unless explicitly overridden by the
+   application) depending on the following:
+
+   o  Requested bitrate
+
+   o  Input sampling rate
+
+   o  Type of signal (speech vs. music)
+
+   o  Frame size in use
+
+   The type of signal currently needs to be provided by the application
+   (though it can be changed in real-time).  An Opus encoder
+   implementation could also do automatic detection, but since Opus is
+   an interactive codec, such an implementation would likely have to
+   either delay the signal (for non-interactive applications) or delay
+   the mode switching decisions (for interactive applications).
+
+
+
+
+Valin, et al.                Standards Track                  [Page 131]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   When the encoder is configured for voice over IP applications, the
+   input signal is filtered by a high-pass filter to remove the lowest
+   part of the spectrum that contains little speech energy and may
+   contain background noise.  This is a second order Auto Regressive
+   Moving Average (i.e., with poles and zeros) filter with a cut-off
+   frequency around 50 Hz.  In the future, a music detector may also be
+   used to lower the cut-off frequency when the input signal is detected
+   to be music rather than speech.
+
+5.1.  Range Encoder
+
+   The range coder acts as the bit-packer for Opus.  It is used in three
+   different ways: to encode
+
+   o  Entropy-coded symbols with a fixed probability model using
+      ec_encode() (entenc.c),
+
+   o  Integers from 0 to (2**M - 1) using ec_enc_uint() or ec_enc_bits()
+      (entenc.c),
+
+   o  Integers from 0 to (ft - 1) (where ft is not a power of two) using
+      ec_enc_uint() (entenc.c).
+
+   The range encoder maintains an internal state vector composed of the
+   four-tuple (val, rng, rem, ext) representing the low end of the
+   current range, the size of the current range, a single buffered
+   output byte, and a count of additional carry-propagating output
+   bytes.  Both val and rng are 32-bit unsigned integer values, rem is a
+   byte value or less than 255 or the special value -1, and ext is an
+   unsigned integer with at least 11 bits.  This state vector is
+   initialized at the start of each frame to the value
+   (0, 2**31, -1, 0).  After encoding a sequence of symbols, the value
+   of rng in the encoder should exactly match the value of rng in the
+   decoder after decoding the same sequence of symbols.  This is a
+   powerful tool for detecting errors in either an encoder or decoder
+   implementation.  The value of val, on the other hand, represents
+   different things in the encoder and decoder, and is not expected to
+   match.
+
+   The decoder has no analog for rem and ext.  These are used to perform
+   carry propagation in the renormalization loop below.  Each iteration
+   of this loop produces 9 bits of output, consisting of 8 data bits and
+   a carry flag.  The encoder cannot determine the final value of the
+   output bytes until it propagates these carry flags.  Therefore, the
+   reference implementation buffers a single non-propagating output byte
+   (i.e., one less than 255) in rem and keeps a count of additional
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 132]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   propagating (i.e., 255) output bytes in ext.  An implementation may
+   choose to use any mathematically equivalent scheme to perform carry
+   propagation.
+
+5.1.1.  Encoding Symbols
+
+   The main encoding function is ec_encode() (entenc.c), which encodes
+   symbol k in the current context using the same three-tuple
+   (fl[k], fh[k], ft) as the decoder to describe the range of the symbol
+   (see Section 4.1).
+
+   ec_encode() updates the state of the encoder as follows.  If fl[k] is
+   greater than zero, then
+
+                                       rng
+                     val = val + rng - --- * (ft - fl)
+                                       ft
+
+                           rng
+                     rng = --- * (fh - fl)
+                           ft
+
+   Otherwise, val is unchanged and
+
+                                    rng
+                        rng = rng - --- * (fh - fl)
+                                    ft
+
+   The divisions here are integer division.
+
+5.1.1.1.  Renormalization
+
+   After this update, the range is normalized using a procedure very
+   similar to that of Section 4.1.2.1, implemented by ec_enc_normalize()
+   (entenc.c).  The following process is repeated until rng > 2**23.
+   First, the top 9 bits of val, (val>>23), are sent to the carry
+   buffer, described in Section 5.1.1.2.  Then, the encoder sets
+
+                        val = (val<<8) & 0x7FFFFFFF
+
+                        rng = rng<<8
+
+5.1.1.2.  Carry Propagation and Output Buffering
+
+   The function ec_enc_carry_out() (entenc.c) implements carry
+   propagation and output buffering.  It takes, as input, a 9-bit
+   unsigned value, c, consisting of 8 data bits and an additional carry
+
+
+
+
+Valin, et al.                Standards Track                  [Page 133]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   bit.  If c is equal to the value 255, then ext is simply incremented,
+   and no other state updates are performed.  Otherwise, let b = (c>>8)
+   be the carry bit.  Then,
+
+   o  If the buffered byte rem contains a value other than -1, the
+      encoder outputs the byte (rem + b).  Otherwise, if rem is -1, no
+      byte is output.
+
+   o  If ext is non-zero, then the encoder outputs ext bytes -- all with
+      a value of 0 if b is set, or 255 if b is unset -- and sets ext to
+      0.
+
+   o  rem is set to the 8 data bits:
+
+                               rem = c & 255
+
+5.1.2.  Alternate Encoding Methods
+
+   The reference implementation uses three additional encoding methods
+   that are exactly equivalent to the above, but make assumptions and
+   simplifications that allow for a more efficient implementation.
+
+5.1.2.1.  ec_encode_bin()
+
+   The first is ec_encode_bin() (entenc.c), defined using the parameter
+   ftb instead of ft.  It is mathematically equivalent to calling
+   ec_encode() with ft = (1<<ftb), but it avoids using division.
+
+5.1.2.2.  ec_enc_bit_logp()
+
+   The next is ec_enc_bit_logp() (entenc.c), which encodes a single
+   binary symbol.  The context is described by a single parameter, logp,
+   which is the absolute value of the base-2 logarithm of the
+   probability of a "1".  It is mathematically equivalent to calling
+   ec_encode() with the 3-tuple (fl[k] = 0, fh[k] = (1<<logp) - 1,
+   ft = (1<<logp)) if k is 0 and with (fl[k] = (1<<logp) - 1,
+   fh[k] = ft = (1<<logp)) if k is 1.  The implementation requires no
+   multiplications or divisions.
+
+5.1.2.3.  ec_enc_icdf()
+
+   The last is ec_enc_icdf() (entenc.c), which encodes a single binary
+   symbol with a table-based context of up to 8 bits.  This uses the
+   same icdf table as ec_dec_icdf() from Section 4.1.3.3.  The function
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 134]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   is mathematically equivalent to calling ec_encode() with
+   fl[k] = (1<<ftb) - icdf[k-1] (or 0 if k == 0), fh[k] = (1<<ftb) -
+    icdf[k], and ft = (1<<ftb).  This only saves a few arithmetic
+   operations over ec_encode_bin(), but it allows the encoder to use the
+   same icdf tables as the decoder.
+
+5.1.3.  Encoding Raw Bits
+
+   The raw bits used by the CELT layer are packed at the end of the
+   buffer using ec_enc_bits() (entenc.c).  Because the raw bits may
+   continue into the last byte output by the range coder if there is
+   room in the low-order bits, the encoder must be prepared to merge
+   these values into a single byte.  The procedure in Section 5.1.5 does
+   this in a way that ensures both the range coded data and the raw bits
+   can be decoded successfully.
+
+5.1.4.  Encoding Uniformly Distributed Integers
+
+   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
+   Section 4.1.5), 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.
+
+   ec_enc_uint() takes a two-tuple (t, ft), where t is the unsigned
+   integer to be encoded, 0 <= t < ft, and ft is not necessarily a power
+   of two.  Let ftb = ilog(ft - 1), i.e., the number of bits required to
+   store (ft - 1) in two's complement notation.  If ftb is 8 or less,
+   then t is encoded directly using ec_encode() with the three-tuple (t,
+   t + 1, ft).
+
+   If ftb is greater than 8, then the top 8 bits of t are encoded using
+   the three-tuple (t>>(ftb - 8), (t>>(ftb - 8)) + 1,
+   ((ft - 1)>>(ftb - 8)) + 1), and the remaining bits,
+   (t & ((1<<(ftb - 8)) - 1), are encoded as raw bits with
+   ec_enc_bits().
+
+5.1.5.  Finalizing the Stream
+
+   After all symbols are encoded, the stream must be finalized by
+   outputting a value inside the current range.  Let end be the unsigned
+   integer in the interval [val, val + rng) with the largest number of
+   trailing zero bits, b, such that (end + (1<<b) - 1) is also in the
+   interval [val, val + rng).  This choice of end allows the maximum
+   number of trailing bits to be set to arbitrary values while still
+   ensuring the range coded part of the buffer can be decoded correctly.
+
+
+
+
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+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Then, while end is not zero, the top 9 bits of end, i.e., (end>>23),
+   are passed to the carry buffer in accordance with the procedure in
+   Section 5.1.1.2, and end is updated via
+
+                        end = (end<<8) & 0x7FFFFFFF
+
+   Finally, if the buffered output byte, rem, is neither zero nor the
+   special value -1, or the carry count, ext, is greater than zero, then
+   9 zero bits are sent to the carry buffer to flush it to the output
+   buffer.  When outputting the final byte from the range coder, if it
+   would overlap any raw bits already packed into the end of the output
+   buffer, they should be ORed into the same byte.  The bit allocation
+   routines in the CELT layer should ensure that this can be done
+   without corrupting the range coder data so long as end is chosen as
+   described above.  If there is any space between the end of the range
+   coder data and the end of the raw bits, it is padded with zero bits.
+   This entire process is implemented by ec_enc_done() (entenc.c).
+
+5.1.6.  Current Bit Usage
+
+   The bit allocation routines in Opus need to be able to determine a
+   conservative upper bound on the number of bits that have been used to
+   encode the current frame thus far.  This drives allocation decisions
+   and ensures that the range coder and raw bits will not overflow the
+   output buffer.  This is computed in the reference implementation to
+   whole-bit precision by the function ec_tell() (entcode.h) and to
+   fractional 1/8th bit precision by the function ec_tell_frac()
+   (entcode.c).  Like all operations in the range coder, it must be
+   implemented in a bit-exact manner, and it must produce exactly the
+   same value returned by the same functions in the decoder after
+   decoding the same symbols.
+
+5.2.  SILK Encoder
+
+   In many respects, the SILK encoder mirrors the SILK decoder described
+   in Section 4.2.  Details such as the quantization and range coder
+   tables can be found there, while this section describes the high-
+   level design choices that were made.  The diagram below shows the
+   basic modules of the SILK encoder.
+
+               +----------+    +--------+    +---------+
+               |  Sample  |    | Stereo |    |  SILK   |
+        ------>|   Rate   |--->| Mixing |--->|  Core   |---------->
+        Input  |Conversion|    |        |    | Encoder |  Bitstream
+               +----------+    +--------+    +---------+
+
+                          Figure 21: SILK Encoder
+
+
+
+
+Valin, et al.                Standards Track                  [Page 136]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+5.2.1.  Sample Rate Conversion
+
+   The input signal's sampling rate is adjusted by a sample rate
+   conversion module so that it matches the SILK internal sampling rate.
+   The input to the sample rate converter is delayed by a number of
+   samples depending on the sample rate ratio, such that the overall
+   delay is constant for all input and output sample rates.
+
+5.2.2.  Stereo Mixing
+
+   The stereo mixer is only used for stereo input signals.  It converts
+   a stereo left-right signal into an adaptive mid-side representation.
+   The first step is to compute non-adaptive mid-side signals as half
+   the sum and difference between left and right signals.  The side
+   signal is then minimized in energy by subtracting a prediction of it
+   based on the mid signal.  This prediction works well when the left
+   and right signals exhibit linear dependency, for instance, for an
+   amplitude-panned input signal.  Like in the decoder, the prediction
+   coefficients are linearly interpolated during the first 8 ms of the
+   frame.  The mid signal is always encoded, whereas the residual side
+   signal is only encoded if it has sufficient energy compared to the
+   mid signal's energy.  If it has not, the "mid_only_flag" is set
+   without encoding the side signal.
+
+   The predictor coefficients are coded regardless of whether the side
+   signal is encoded.  For each frame, two predictor coefficients are
+   computed, one that predicts between low-passed mid and side channels,
+   and one that predicts between high-passed mid and side channels.  The
+   low-pass filter is a simple three-tap filter and creates a delay of
+   one sample.  The high-pass filtered signal is the difference between
+   the mid signal delayed by one sample and the low-passed signal.
+   Instead of explicitly computing the high-passed signal, it is
+   computationally more efficient to transform the prediction
+   coefficients before applying them to the filtered mid signal, as
+   follows:
+
+               pred(n) = LP(n) * w0 + HP(n) * w1
+                       = LP(n) * w0 + (mid(n-1) - LP(n)) * w1
+                       = LP(n) * (w0 - w1) + mid(n-1) * w1
+
+
+   where w0 and w1 are the low-pass and high-pass prediction
+   coefficients, mid(n-1) is the mid signal delayed by one sample, LP(n)
+   and HP(n) are the low-passed and high-passed signals and pred(n) is
+   the prediction signal that is subtracted from the side signal.
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 137]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+5.2.3.  SILK Core Encoder
+
+   What follows is a description of the core encoder and its components.
+   For simplicity, the core encoder is referred to simply as the encoder
+   in the remainder of this section.  An overview of the encoder is
+   given in Figure 22.
+
+                                                                +---+
+                             +--------------------------------->|   |
+        +---------+          |      +---------+                 |   |
+        |Voice    |          |      |LTP      |12               |   |
+    +-->|Activity |--+       +----->|Scaling  |-----------+---->|   |
+    |   |Detection|3 |       |      |Control  |<--+       |     |   |
+    |   +---------+  |       |      +---------+   |       |     |   |
+    |                |       |      +---------+   |       |     |   |
+    |                |       |      |Gains    |   |       |     |   |
+    |                |       |  +-->|Processor|---|---+---|---->| R |
+    |                |       |  |   |         |11 |   |   |     | a |
+    |               \/       |  |   +---------+   |   |   |     | n |
+    |          +---------+   |  |   +---------+   |   |   |     | g |
+    |          |Pitch    |   |  |   |LSF      |   |   |   |     | e |
+    |       +->|Analysis |---+  |   |Quantizer|---|---|---|---->|   |
+    |       |  |         |4  |  |   |         |8  |   |   |     | E |-->
+    |       |  +---------+   |  |   +---------+   |   |   |     | n | 2
+    |       |                |  |    9/\  10|     |   |   |     | c |
+    |       |                |  |     |    \/     |   |   |     | o |
+    |       |  +---------+   |  |   +----------+  |   |   |     | d |
+    |       |  |Noise    |   +--|-->|Prediction|--+---|---|---->| e |
+    |       +->|Shaping  |---|--+   |Analysis  |7 |   |   |     | r |
+    |       |  |Analysis |5  |  |   |          |  |   |   |     |   |
+    |       |  +---------+   |  |   +----------+  |   |   |     |   |
+    |       |                |  |        /\       |   |   |     |   |
+    |       |     +----------|--|--------+        |   |   |     |   |
+    |       |     |         \/  \/               \/  \/  \/     |   |
+    |       |     |      +----------+          +------------+   |   |
+    |       |     |      |          |          |Noise       |   |   |
+   -+-------+-----+----->|Pre-filter|--------->|Shaping     |-->|   |
+   1                     |          | 6        |Quantization|13 |   |
+                         +----------+          +------------+   +---+
+
+   1:  Input speech signal
+   2:  Range encoded bitstream
+   3:  Voice activity estimate
+   4:  Pitch lags (per 5 ms) and voicing decision (per 20 ms)
+   5:  Noise shaping quantization coefficients
+     - Short-term synthesis and analysis
+       noise shaping coefficients (per 5 ms)
+
+
+
+
+Valin, et al.                Standards Track                  [Page 138]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+     - Long-term synthesis and analysis noise
+       shaping coefficients (per 5 ms and for voiced speech only)
+     - Noise shaping tilt (per 5 ms)
+     - Quantizer gain/step size (per 5 ms)
+   6:  Input signal filtered with analysis noise shaping filters
+   7:  Short- and Long-Term Prediction coefficients
+       LTP (per 5 ms) and LPC (per 20 ms)
+   8:  LSF quantization indices
+   9:  LSF coefficients
+   10: Quantized LSF coefficients
+   11: Processed gains, and synthesis noise shape coefficients
+   12: LTP state scaling coefficient.  Controlling error
+       propagation / prediction gain trade-off
+   13: Quantized signal
+
+                       Figure 22: SILK Core Encoder
+
+5.2.3.1.  Voice Activity Detection
+
+   The input signal is processed by a Voice Activity Detection (VAD)
+   algorithm to produce a measure of voice activity, spectral tilt, and
+   signal-to-noise estimates for each frame.  The VAD uses a sequence of
+   half-band filterbanks to split the signal into four subbands:
+   0...Fs/16, Fs/16...Fs/8, Fs/8...Fs/4, and Fs/4...Fs/2, where Fs is
+   the sampling frequency (8, 12, 16, or 24 kHz).  The lowest subband,
+   from 0 - Fs/16, is high-pass filtered with a first-order moving
+   average (MA) filter (with transfer function H(z) = 1-z**(-1)) to
+   reduce the energy at the lowest frequencies.  For each frame, the
+   signal energy per subband is computed.  In each subband, a noise
+   level estimator tracks the background noise level and a Signal-to-
+   Noise Ratio (SNR) value is computed as the logarithm of the ratio of
+   energy-to-noise level.  Using these intermediate variables, the
+   following parameters are calculated for use in other SILK modules:
+
+   o  Average SNR.  The average of the subband SNR values.
+
+   o  Smoothed subband SNRs.  Temporally smoothed subband SNR values.
+
+   o  Speech activity level.  Based on the average SNR and a weighted
+      average of the subband energies.
+
+   o  Spectral tilt.  A weighted average of the subband SNRs, with
+      positive weights for the low subbands and negative weights for the
+      high subbands.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 139]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+5.2.3.2.  Pitch Analysis
+
+   The input signal is processed by the open loop pitch estimator shown
+   in Figure 23.
+
+                                    +--------+  +----------+
+                                    |2 x Down|  |Time-     |
+                                 +->|sampling|->|Correlator|     |
+                                 |  |        |  |          |     |4
+                                 |  +--------+  +----------+    \/
+                                 |                    | 2    +-------+
+                                 |                    |  +-->|Speech |5
+       +---------+    +--------+ |                   \/  |   |Type   |->
+       |LPC      |    |Down    | |              +----------+ |       |
+    +->|Analysis | +->|sample  |-+------------->|Time-     | +-------+
+    |  |         | |  |to 8 kHz|                |Correlator|----------->
+    |  +---------+ |  +--------+                |__________|          6
+    |       |      |                                  |3
+    |      \/      |                                 \/
+    |  +---------+ |                            +----------+
+    |  |Whitening| |                            |Time-     |
+   -+->|Filter   |-+--------------------------->|Correlator|----------->
+   1   |         |                              |          |          7
+       +---------+                              +----------+
+
+   1: Input signal
+   2: Lag candidates from stage 1
+   3: Lag candidates from stage 2
+   4: Correlation threshold
+   5: Voiced/unvoiced flag
+   6: Pitch correlation
+   7: Pitch lags
+
+              Figure 23: Block Diagram of the Pitch Estimator
+
+   The pitch analysis finds a binary voiced/unvoiced classification,
+   and, for frames classified as voiced, four pitch lags per frame --
+   one for each 5 ms subframe -- and a pitch correlation indicating the
+   periodicity of the signal.  The input is first whitened using a
+   Linear Prediction (LP) whitening filter, where the coefficients are
+   computed through standard Linear Predictive Coding (LPC) analysis.
+   The order of the whitening filter is 16 for best results, but is
+   reduced to 12 for medium complexity and 8 for low complexity modes.
+   The whitened signal is analyzed to find pitch lags for which the time
+   correlation is high.  The analysis consists of three stages for
+   reducing the complexity:
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 140]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  In the first stage, the whitened signal is downsampled to 4 kHz
+      (from 8 kHz), and the current frame is correlated to a signal
+      delayed by a range of lags, starting from a shortest lag
+      corresponding to 500 Hz, to a longest lag corresponding to 56 Hz.
+
+   o  The second stage operates on an 8 kHz signal (downsampled from 12,
+      16, or 24 kHz) and measures time correlations only near the lags
+      corresponding to those that had sufficiently high correlations in
+      the first stage.  The resulting correlations are adjusted for a
+      small bias towards short lags to avoid ending up with a multiple
+      of the true pitch lag.  The highest adjusted correlation is
+      compared to a threshold depending on:
+
+      *  Whether the previous frame was classified as voiced.
+
+      *  The speech activity level.
+
+      *  The spectral tilt.
+
+      If the threshold is exceeded, the current frame is classified as
+      voiced and the lag with the highest adjusted correlation is stored
+      for a final pitch analysis of the highest precision in the third
+      stage.
+
+   o  The last stage operates directly on the whitened input signal to
+      compute time correlations for each of the four subframes
+      independently in a narrow range around the lag with highest
+      correlation from the second stage.
+
+5.2.3.3.  Noise Shaping Analysis
+
+   The noise shaping analysis finds gains and filter coefficients used
+   in the pre-filter and noise shaping quantizer.  These parameters are
+   chosen such that they will fulfill several requirements:
+
+   o  Balancing quantization noise and bitrate.  The quantization gains
+      determine the step size between reconstruction levels of the
+      excitation signal.  Therefore, increasing the quantization gain
+      amplifies quantization noise, but also reduces the bitrate by
+      lowering the entropy of the quantization indices.
+
+   o  Spectral shaping of the quantization noise; the noise shaping
+      quantizer is capable of reducing quantization noise in some parts
+      of the spectrum at the cost of increased noise in other parts
+      without substantially changing the bitrate.  By shaping the noise
+      such that it follows the signal spectrum, it becomes less audible.
+      In practice, best results are obtained by making the shape of the
+      noise spectrum slightly flatter than the signal spectrum.
+
+
+
+Valin, et al.                Standards Track                  [Page 141]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   o  De-emphasizing spectral valleys; by using different coefficients
+      in the analysis and synthesis part of the pre-filter and noise
+      shaping quantizer, the levels of the spectral valleys can be
+      decreased relative to the levels of the spectral peaks such as
+      speech formants and harmonics.  This reduces the entropy of the
+      signal, which is the difference between the coded signal and the
+      quantization noise, thus lowering the bitrate.
+
+   o  Matching the levels of the decoded speech formants to the levels
+      of the original speech formants; an adjustment gain and a first
+      order tilt coefficient are computed to compensate for the effect
+      of the noise shaping quantization on the level and spectral tilt.
+
+                 / \   ___
+                  |   // \\
+                  |  //   \\     ____
+                  |_//     \\___//  \\         ____
+                  | /  ___  \   /    \\       //  \\
+                P |/  /   \  \_/      \\_____//    \\
+                o |  /     \     ____  \     /      \\
+                w | /       \___/    \  \___/  ____  \\___ 1
+                e |/                  \       /    \  \
+                r |                    \_____/      \  \__ 2
+                  |                                  \
+                  |                                   \___ 3
+                  |
+                  +---------------------------------------->
+                                   Frequency
+
+               1: Input signal spectrum
+               2: De-emphasized and level matched spectrum
+               3: Quantization noise spectrum
+
+      Figure 24: Noise Shaping and Spectral De-emphasis Illustration
+
+   Figure 24 shows an example of an input signal spectrum (1).  After
+   de-emphasis and level matching, the spectrum has deeper valleys (2).
+   The quantization noise spectrum (3) more or less follows the input
+   signal spectrum, while having slightly less pronounced peaks.  The
+   entropy, which provides a lower bound on the bitrate for encoding the
+   excitation signal, is proportional to the area between the de-
+   emphasized spectrum (2) and the quantization noise spectrum (3).
+   Without de-emphasis, the entropy is proportional to the area between
+   input spectrum (1) and quantization noise (3) -- clearly higher.
+
+   The transformation from input signal to de-emphasized signal can be
+   described as a filtering operation with a filter
+
+
+
+
+Valin, et al.                Standards Track                  [Page 142]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+                                             -1    Wana(z)
+                  H(z) = G * ( 1 - c_tilt * z  ) * -------
+                                                   Wsyn(z)
+
+   having an adjustment gain G, a first order tilt adjustment filter
+   with tilt coefficient c_tilt, and where
+
+                       16                           d
+                       __            -k        -L  __            -k
+        Wana(z) = (1 - \ a_ana(k) * z  )*(1 - z  * \ b_ana(k) * z  )
+                       /_                          /_
+                       k=1                         k=-d
+
+   is the analysis part of the de-emphasis filter, consisting of the
+   short-term shaping filter with coefficients a_ana(k) and the long-
+   term shaping filter with coefficients b_ana(k) and pitch lag L.  The
+   parameter d determines the number of long-term shaping filter taps.
+
+   Similarly, but without the tilt adjustment, the synthesis part can be
+   written as
+
+                       16                           d
+                       __            -k        -L  __            -k
+        Wsyn(z) = (1 - \ a_syn(k) * z  )*(1 - z  * \ b_syn(k) * z  )
+                       /_                          /_
+                       k=1                         k=-d
+
+   All noise shaping parameters are computed and applied per subframe of
+   5 ms.  First, an LPC analysis is performed on a windowed signal block
+   of 15 ms.  The signal block has a look-ahead of 5 ms relative to the
+   current subframe, and the window is an asymmetric sine window.  The
+   LPC analysis is done with the autocorrelation method, with an order
+   of between 8, in lowest-complexity mode, and 16, for best quality.
+
+   Optionally, the LPC analysis and noise shaping filters are warped by
+   replacing the delay elements by first-order allpass filters.  This
+   increases the frequency resolution at low frequencies and reduces it
+   at high ones, which better matches the human auditory system and
+   improves quality.  The warped analysis and filtering comes at a cost
+   in complexity and is therefore only done in higher complexity modes.
+
+   The quantization gain is found by taking the square root of the
+   residual energy from the LPC analysis and multiplying it by a value
+   inversely proportional to the coding quality control parameter and
+   the pitch correlation.
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 143]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   Next, the two sets of short-term noise shaping coefficients a_ana(k)
+   and a_syn(k) are obtained by applying different amounts of bandwidth
+   expansion to the coefficients found in the LPC analysis.  This
+   bandwidth expansion moves the roots of the LPC polynomial towards the
+   origin, using the formulas
+
+                                              k
+                         a_ana(k) = a(k)*g_ana   and
+
+                                              k
+                         a_syn(k) = a(k)*g_syn
+
+   where a(k) is the k'th LPC coefficient, and the bandwidth expansion
+   factors g_ana and g_syn are calculated as
+
+                         g_ana = 0.95 - 0.01*C  and
+
+                         g_syn = 0.95 + 0.01*C
+
+   where C is the coding quality control parameter between 0 and 1.
+   Applying more bandwidth expansion to the analysis part than to the
+   synthesis part gives the desired de-emphasis of spectral valleys in
+   between formants.
+
+   The long-term shaping is applied only during voiced frames.  It uses
+   a three-tap filter, described by
+
+                   b_ana = F_ana * [0.25, 0.5, 0.25]  and
+
+                   b_syn = F_syn * [0.25, 0.5, 0.25].
+
+   For unvoiced frames, these coefficients are set to 0.  The
+   multiplication factors F_ana and F_syn are chosen between 0 and 1,
+   depending on the coding quality control parameter, as well as the
+   calculated pitch correlation and smoothed subband SNR of the lowest
+   subband.  By having F_ana less than F_syn, the pitch harmonics are
+   emphasized relative to the valleys in between the harmonics.
+
+   The tilt coefficient c_tilt is for unvoiced frames chosen as
+
+                               c_tilt = 0.25
+
+   and as
+
+                         c_tilt = 0.25 + 0.2625 * V
+
+   for voiced frames, where V is the voice activity level between 0 and
+   1.
+
+
+
+Valin, et al.                Standards Track                  [Page 144]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   The adjustment gain G serves to correct any level mismatch between
+   the original and decoded signals that might arise from the noise
+   shaping and de-emphasis.  This gain is computed as the ratio of the
+   prediction gain of the short-term analysis and synthesis filter
+   coefficients.  The prediction gain of an LPC synthesis filter is the
+   square root of the output energy when the filter is excited by a
+   unit-energy impulse on the input.  An efficient way to compute the
+   prediction gain is by first computing the reflection coefficients
+   from the LPC coefficients through the step-down algorithm, and
+   extracting the prediction gain from the reflection coefficients as
+
+                                    K
+                                   ___          2  -0.5
+                      predGain = ( | | 1 - (r_k)  )
+                                   k=1
+
+   where r_k is the k'th reflection coefficient.
+
+   Initial values for the quantization gains are computed as the square
+   root of the residual energy of the LPC analysis, adjusted by the
+   coding quality control parameter.  These quantization gains are later
+   adjusted based on the results of the prediction analysis.
+
+5.2.3.4.  Prediction Analysis
+
+   The prediction analysis is performed in one of two ways depending on
+   how the pitch estimator classified the frame.  The processing for
+   voiced and unvoiced speech is described in Section 5.2.3.4.1 and
+   Section 5.2.3.4.2, respectively.  Inputs to this function include the
+   pre-whitened signal from the pitch estimator (see Section 5.2.3.2).
+
+5.2.3.4.1.  Voiced Speech
+
+   For a frame of voiced speech, the pitch pulses will remain dominant
+   in the pre-whitened input signal.  Further whitening is desirable as
+   it leads to higher quality at the same available bitrate.  To achieve
+   this, a Long-Term Prediction (LTP) analysis is carried out to
+   estimate the coefficients of a fifth-order LTP filter for each of
+   four subframes.  The LTP coefficients are quantized using the method
+   described in Section 5.2.3.6, and the quantized LTP coefficients are
+   used to compute the LTP residual signal.  This LTP residual signal is
+   the input to an LPC analysis where the LPC coefficients are estimated
+   using Burg's method [BURG], such that the residual energy is
+   minimized.  The estimated LPC coefficients are converted to a Line
+   Spectral Frequency (LSF) vector and quantized as described in
+   Section 5.2.3.5.  After quantization, the quantized LSF vector is
+   converted back to LPC coefficients using the full procedure in
+   Section 4.2.7.5.  By using quantized LTP coefficients and LPC
+
+
+
+Valin, et al.                Standards Track                  [Page 145]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   coefficients derived from the quantized LSF coefficients, the encoder
+   remains fully synchronized with the decoder.  The quantized LPC and
+   LTP coefficients are also used to filter the input signal and measure
+   residual energy for each of the four subframes.
+
+5.2.3.4.2.  Unvoiced Speech
+
+   For a speech signal that has been classified as unvoiced, there is no
+   need for LTP filtering, as it has already been determined that the
+   pre-whitened input signal is not periodic enough within the allowed
+   pitch period range for LTP analysis to be worth the cost in terms of
+   complexity and bitrate.  The pre-whitened input signal is therefore
+   discarded, and, instead, the input signal is used for LPC analysis
+   using Burg's method.  The resulting LPC coefficients are converted to
+   an LSF vector and quantized as described in the following section.
+   They are then transformed back to obtain quantized LPC coefficients,
+   which are then used to filter the input signal and measure residual
+   energy for each of the four subframes.
+
+5.2.3.4.2.1.  Burg's Method
+
+   The main purpose of linear prediction in SILK is to reduce the
+   bitrate by minimizing the residual energy.  At least at high
+   bitrates, perceptual aspects are handled independently by the noise
+   shaping filter.  Burg's method is used because it provides higher
+   prediction gain than the autocorrelation method and, unlike the
+   covariance method, produces stable filters (assuming numerical errors
+   don't spoil that).  SILK's implementation of Burg's method is also
+   computationally faster than the autocovariance method.  The
+   implementation of Burg's method differs from traditional
+   implementations in two aspects.  The first difference is that it
+   operates on autocorrelations, similar to the Schur algorithm [SCHUR],
+   but with a simple update to the autocorrelations after finding each
+   reflection coefficient to make the result identical to Burg's method.
+   This brings down the complexity of Burg's method to near that of the
+   autocorrelation method.  The second difference is that the signal in
+   each subframe is scaled by the inverse of the residual quantization
+   step size.  Subframes with a small quantization step size will, on
+   average, spend more bits for a given amount of residual energy than
+   subframes with a large step size.  Without scaling, Burg's method
+   minimizes the total residual energy in all subframes, which doesn't
+   necessarily minimize the total number of bits needed for coding the
+   quantized residual.  The residual energy of the scaled subframes is a
+   better measure for that number of bits.
+
+
+
+
+
+
+
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+
+
+5.2.3.5.  LSF Quantization
+
+   Unlike many other speech codecs, SILK uses variable bitrate coding
+   for the LSFs.  This improves the average rate-distortion (R-D) trade-
+   off and reduces outliers.  The variable bitrate coding minimizes a
+   linear combination of the weighted quantization errors and the
+   bitrate.  The weights for the quantization errors are the Inverse
+   Harmonic Mean Weighting (IHMW) function proposed by Laroia et al.
+   (see [LAROIA-ICASSP]).  These weights are referred to here as Laroia
+   weights.
+
+   The LSF quantizer consists of two stages.  The first stage is an
+   (unweighted) vector quantizer (VQ), with a codebook size of 32
+   vectors.  The quantization errors for the codebook vector are sorted,
+   and for the N best vectors a second stage quantizer is run.  By
+   varying the number N, a trade-off is made between R-D performance and
+   computational efficiency.  For each of the N codebook vectors, the
+   Laroia weights corresponding to that vector (and not to the input
+   vector) are calculated.  Then, the residual between the input LSF
+   vector and the codebook vector is scaled by the square roots of these
+   Laroia weights.  This scaling partially normalizes error sensitivity
+   for the residual vector so that a uniform quantizer with fixed step
+   sizes can be used in the second stage without too much performance
+   loss.  Additionally, by scaling with Laroia weights determined from
+   the first-stage codebook vector, the process can be reversed in the
+   decoder.
+
+   The second stage uses predictive delayed decision scalar
+   quantization.  The quantization error is weighted by Laroia weights
+   determined from the LSF input vector.  The predictor multiplies the
+   previous quantized residual value by a prediction coefficient that
+   depends on the vector index from the first stage VQ and on the
+   location in the LSF vector.  The prediction is subtracted from the
+   LSF residual value before quantizing the result and is added back
+   afterwards.  This subtraction can be interpreted as shifting the
+   quantization levels of the scalar quantizer, and as a result the
+   quantization error of each value depends on the quantization decision
+   of the previous value.  This dependency is exploited by the delayed
+   decision mechanism to search for a quantization sequency with best
+   R-D performance with a Viterbi-like algorithm [VITERBI].  The
+   quantizer processes the residual LSF vector in reverse order (i.e.,
+   it starts with the highest residual LSF value).  This is done because
+   the prediction works slightly better in the reverse direction.
+
+
+
+
+
+
+
+
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+
+
+   The quantization index of the first stage is entropy coded.  The
+   quantization sequence from the second stage is also entropy coded,
+   where for each element the probability table is chosen depending on
+   the vector index from the first stage and the location of that
+   element in the LSF vector.
+
+5.2.3.5.1.  LSF Stabilization
+
+   If the input is stable, finding the best candidate usually results in
+   a quantized vector that is also stable.  Because of the two-stage
+   approach, however, it is possible that the best quantization
+   candidate is unstable.  The encoder applies the same stabilization
+   procedure applied by the decoder (see Section 4.2.7.5.4) to ensure
+   the LSF parameters are within their valid range, increasingly sorted,
+   and have minimum distances between each other and the border values.
+
+5.2.3.6.  LTP Quantization
+
+   For voiced frames, the prediction analysis described in
+   Section 5.2.3.4.1 resulted in four sets (one set per subframe) of
+   five LTP coefficients, plus four weighting matrices.  The LTP
+   coefficients for each subframe are quantized using entropy
+   constrained vector quantization.  A total of three vector codebooks
+   are available for quantization, with different rate-distortion trade-
+   offs.  The three codebooks have 10, 20, and 40 vectors and average
+   rates of about 3, 4, and 5 bits per vector, respectively.
+   Consequently, the first codebook has larger average quantization
+   distortion at a lower rate, whereas the last codebook has smaller
+   average quantization distortion at a higher rate.  Given the
+   weighting matrix W_ltp and LTP vector b, the weighted rate-distortion
+   measure for a codebook vector cb_i with rate r_i is give by
+
+               RD = u * (b - cb_i)' * W_ltp * (b - cb_i) + r_i
+
+   where u is a fixed, heuristically determined parameter balancing the
+   distortion and rate.  Which codebook gives the best performance for a
+   given LTP vector depends on the weighting matrix for that LTP vector.
+   For example, for a low valued W_ltp, it is advantageous to use the
+   codebook with 10 vectors as it has a lower average rate.  For a large
+   W_ltp, on the other hand, it is often better to use the codebook with
+   40 vectors, as it is more likely to contain the best codebook vector.
+   The weighting matrix W_ltp depends mostly on two aspects of the input
+   signal.  The first is the periodicity of the signal; the more
+   periodic, the larger W_ltp.  The second is the change in signal
+   energy in the current subframe, relative to the signal one pitch lag
+   earlier.  A decaying energy leads to a larger W_ltp than an
+   increasing energy.  Both aspects fluctuate relatively slowly, which
+   causes the W_ltp matrices for different subframes of one frame often
+
+
+
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+
+
+   to be similar.  Because of this, one of the three codebooks typically
+   gives good performance for all subframes.  Therefore, the codebook
+   search for the subframe LTP vectors is constrained to only allow
+   codebook vectors to be chosen from the same codebook, resulting in a
+   rate reduction.
+
+   To find the best codebook, each of the three vector codebooks is used
+   to quantize all subframe LTP vectors and produce a combined weighted
+   rate-distortion measure for each vector codebook.  The vector
+   codebook with the lowest combined rate-distortion over all subframes
+   is chosen.  The quantized LTP vectors are used in the noise shaping
+   quantizer, and the index of the codebook plus the four indices for
+   the four subframe codebook vectors are passed on to the range
+   encoder.
+
+5.2.3.7.  Pre-filter
+
+   In the pre-filter, the input signal is filtered using the spectral
+   valley de-emphasis filter coefficients from the noise shaping
+   analysis (see Section 5.2.3.3).  By applying only the noise shaping
+   analysis filter to the input signal, it provides the input to the
+   noise shaping quantizer.
+
+5.2.3.8.  Noise Shaping Quantizer
+
+   The noise shaping quantizer independently shapes the signal and
+   coding noise spectra to obtain a perceptually higher quality at the
+   same bitrate.
+
+   The pre-filter output signal is multiplied with a compensation gain G
+   computed in the noise shaping analysis.  Then, the output of a
+   synthesis shaping filter is added, and the output of a prediction
+   filter is subtracted to create a residual signal.  The residual
+   signal is multiplied by the inverse quantized quantization gain from
+   the noise shaping analysis and input to a scalar quantizer.  The
+   quantization indices of the scalar quantizer represent a signal of
+   pulses that is input to the pyramid range encoder.  The scalar
+   quantizer also outputs a quantization signal, which is multiplied by
+   the quantized quantization gain from the noise shaping analysis to
+   create an excitation signal.  The output of the prediction filter is
+   added to the excitation signal to form the quantized output signal
+   y(n).  The quantized output signal y(n) is input to the synthesis
+   shaping and prediction filters.
+
+
+
+
+
+
+
+
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+
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+
+
+   Optionally, the noise shaping quantizer operates in a delayed
+   decision mode.  In this mode, it uses a Viterbi algorithm to keep
+   track of multiple rounding choices in the quantizer and select the
+   best one after a delay of 32 samples.  This improves the rate/
+   distortion performance of the quantizer.
+
+5.2.3.9.  Constant Bitrate Mode
+
+   SILK was designed to run in variable bitrate (VBR) mode.  However,
+   the reference implementation also has a constant bitrate (CBR) mode
+   for SILK.  In CBR mode, SILK will attempt to encode each packet with
+   no more than the allowed number of bits.  The Opus wrapper code then
+   pads the bitstream if any unused bits are left in SILK mode, or it
+   encodes the high band with the remaining number of bits in Hybrid
+   mode.  The number of payload bits is adjusted by changing the
+   quantization gains and the rate/distortion trade-off in the noise
+   shaping quantizer, in an iterative loop around the noise shaping
+   quantizer and entropy coding.  Compared to the SILK VBR mode, the CBR
+   mode has lower audio quality at a given average bitrate and has
+   higher computational complexity.
+
+5.3.  CELT Encoder
+
+   Most of the aspects of the CELT encoder can be directly derived from
+   the description of the decoder.  For example, the filters and
+   rotations in the encoder are simply the inverse of the operation
+   performed by the decoder.  Similarly, the quantizers generally
+   optimize for the mean square error (because noise shaping is part of
+   the bitstream itself), so no special search is required.  For this
+   reason, only the less straightforward aspects of the encoder are
+   described here.
+
+5.3.1.  Pitch Pre-filter
+
+   The pitch pre-filter is applied after the pre-emphasis.  It is
+   applied in such a way as to be the inverse of the decoder's post-
+   filter.  The main non-obvious aspect of the pre-filter is the
+   selection of the pitch period.  The pitch search should be optimized
+   for the following criteria:
+
+   o  continuity: it is important that the pitch period does not change
+      abruptly between frames; and
+
+   o  avoidance of pitch multiples: when the period used is a multiple
+      of the real period (lower frequency fundamental), the post-filter
+      loses most of its ability to reduce noise
+
+
+
+
+
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+
+
+5.3.2.  Bands and Normalization
+
+   The MDCT output is divided into bands that are designed to match the
+   ear's critical bands for the smallest (2.5 ms) frame size.  The
+   larger frame sizes use integer multiples of the 2.5 ms layout.  For
+   each band, the encoder computes the energy that will later be
+   encoded.  Each band is then normalized by the square root of the
+   *unquantized* energy, such that each band now forms a unit vector X.
+   The energy and the normalization are computed by
+   compute_band_energies() and normalise_bands() (bands.c),
+   respectively.
+
+5.3.3.  Energy Envelope Quantization
+
+   Energy quantization (both coarse and fine) can be easily understood
+   from the decoding process.  For all useful bitrates, the coarse
+   quantizer always chooses the quantized log energy value that
+   minimizes the error for each band.  Only at very low rate does the
+   encoder allow larger errors to minimize the rate and avoid using more
+   bits than are available.  When the available CPU requirements allow
+   it, it is best to try encoding the coarse energy both with and
+   without inter-frame prediction such that the best prediction mode can
+   be selected.  The optimal mode depends on the coding rate, the
+   available bitrate, and the current rate of packet loss.
+
+   The fine energy quantizer always chooses the quantized log energy
+   value that minimizes the error for each band because the rate of the
+   fine quantization depends only on the bit allocation and not on the
+   values that are coded.
+
+5.3.4.  Bit Allocation
+
+   The encoder must use exactly the same bit allocation process as used
+   by the decoder and described in Section 4.3.3.  The three mechanisms
+   that can be used by the encoder to adjust the bitrate on a frame-by-
+   frame basis are band boost, allocation trim, and band skipping.
+
+5.3.4.1.  Band Boost
+
+   The reference encoder makes a decision to boost a band when the
+   energy of that band is significantly higher than that of the
+   neighboring bands.  Let E_j be the log-energy of band j, we define
+
+      D_j = 2*E_j - E_j-1 - E_j+1
+
+   The allocation of band j is boosted once if D_j > t1 and twice if D_j
+   > t2.  For LM>=1, t1=2 and t2=4, while for LM<1, t1=3 and t2=5.
+
+
+
+
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+
+
+5.3.4.2.  Allocation Trim
+
+   The allocation trim is a value between 0 and 10 (inclusively) that
+   controls the allocation balance between the low and high frequencies.
+   The encoder starts with a safe "default" of 5 and deviates from that
+   default in two different ways.  First, the trim can deviate by +/- 2
+   depending on the spectral tilt of the input signal.  For signals with
+   more low frequencies, the trim is increased by up to 2, while for
+   signals with more high frequencies, the trim is decreased by up to 2.
+   For stereo inputs, the trim value can be decreased by up to 4 when
+   the inter-channel correlation at low frequency (first 8 bands) is
+   high.
+
+5.3.4.3.  Band Skipping
+
+   The encoder uses band skipping to ensure that the shape of the bands
+   is only coded if there is at least 1/2 bit per sample available for
+   the PVQ.  If not, then no bit is allocated and folding is used
+   instead.  To ensure continuity in the allocation, some amount of
+   hysteresis is added to the process, such that a band that received
+   PVQ bits in the previous frame only needs 7/16 bit/sample to be coded
+   for the current frame, while a band that did not receive PVQ bits in
+   the previous frames needs at least 9/16 bit/sample to be coded.
+
+5.3.5.  Stereo Decisions
+
+   Because CELT applies mid-side stereo coupling in the normalized
+   domain, it does not suffer from important stereo image problems even
+   when the two channels are completely uncorrelated.  For this reason,
+   it is always safe to use stereo coupling on any audio frame.  That
+   being said, there are some frames for which dual (independent) stereo
+   is still more efficient.  This decision is made by comparing the
+   estimated entropy with and without coupling over the first 13 bands,
+   taking into account the fact that all bands with more than two MDCT
+   bins require one extra degree of freedom when coded in mid-side.  Let
+   L1_ms and L1_lr be the L1-norm of the mid-side vector and the L1-norm
+   of the left-right vector, respectively.  The decision to use mid-side
+   is made if and only if
+
+                            L1_ms          L1_lr
+                           --------    <   -----
+                           bins + E        bins
+
+   where bins is the number of MDCT bins in the first 13 bands and E is
+   the number of extra degrees of freedom for mid-side coding.  For
+   LM>1, E=13, otherwise E=5.
+
+
+
+
+
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+
+
+   The reference encoder decides on the intensity stereo threshold based
+   on the bitrate alone.  After taking into account the frame size by
+   subtracting 80 bits per frame for coarse energy, the first band using
+   intensity coding is as follows:
+
+                     +------------------+------------+
+                     | bitrate (kbit/s) | start band |
+                     +------------------+------------+
+                     |        <35       |      8     |
+                     |                  |            |
+                     |       35-50      |     12     |
+                     |                  |            |
+                     |       50-68      |     16     |
+                     |                  |            |
+                     |       84-84      |     18     |
+                     |                  |            |
+                     |      84-102      |     19     |
+                     |                  |            |
+                     |      102-130     |     20     |
+                     |                  |            |
+                     |       >130       |  disabled  |
+                     +------------------+------------+
+
+                 Table 66: Thresholds for Intensity Stereo
+
+5.3.6.  Time-Frequency Decision
+
+   The choice of time-frequency resolution used in Section 4.3.4.5 is
+   based on R-D optimization.  The distortion is the L1-norm (sum of
+   absolute values) of each band after each TF resolution under
+   consideration.  The L1 norm is used because it represents the entropy
+   for a Laplacian source.  The number of bits required to code a change
+   in TF resolution between two bands is higher than the cost of having
+   those two bands use the same resolution, which is what requires the
+   R-D optimization.  The optimal decision is computed using the Viterbi
+   algorithm.  See tf_analysis() in celt/celt.c.
+
+5.3.7.  Spreading Values Decision
+
+   The choice of the spreading value in Table 59 has an impact on the
+   nature of the coding noise introduced by CELT.  The larger the f_r
+   value, the lower the impact of the rotation, and the more tonal the
+   coding noise.  The more tonal the signal, the more tonal the noise
+   should be, so the CELT encoder determines the optimal value for f_r
+   by estimating how tonal the signal is.  The tonality estimate is
+   based on discrete pdf (4-bin histogram) of each band.  Bands that
+
+
+
+
+
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+
+
+   have a large number of small values are considered more tonal and a
+   decision is made by combining all bands with more than 8 samples.
+   See spreading_decision() in celt/bands.c.
+
+5.3.8.  Spherical Vector Quantization
+
+   CELT uses a Pyramid Vector Quantizer (PVQ) [PVQ] for quantizing the
+   details of the spectrum in each band that have not been predicted by
+   the pitch predictor.  The PVQ codebook consists of all sums of K
+   signed pulses in a vector of N samples, where two pulses at the same
+   position are required to have the same sign.  Thus, the codebook
+   includes all integer codevectors y of N dimensions that satisfy
+   sum(abs(y(j))) = K.
+
+   In bands where there are sufficient bits allocated, PVQ is used to
+   encode the unit vector that results from the normalization in
+   Section 5.3.2 directly.  Given a PVQ codevector y, the unit vector X
+   is obtained as X = y/||y||, where ||.|| denotes the L2 norm.
+
+5.3.8.1.  PVQ Search
+
+   The search for the best codevector y is performed by alg_quant()
+   (vq.c).  There are several possible approaches to the search, with a
+   trade-off between quality and complexity.  The method used in the
+   reference implementation computes an initial codeword y1 by
+   projecting the normalized spectrum X onto the codebook pyramid of K-1
+   pulses:
+
+   y0 = truncate_towards_zero( (K-1) * X / sum(abs(X)))
+
+   Depending on N, K and the input data, the initial codeword y0 may
+   contain from 0 to K-1 non-zero values.  All the remaining pulses,
+   with the exception of the last one, are found iteratively with a
+   greedy search that minimizes the normalized correlation between y and
+   X:
+
+                                   T
+                             J = -X * y / ||y||
+
+   The search described above is considered to be a good trade-off
+   between quality and computational cost.  However, there are other
+   possible ways to search the PVQ codebook and the implementers MAY use
+   any other search methods.  See alg_quant() in celt/vq.c.
+
+
+
+
+
+
+
+
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+
+
+5.3.8.2.  PVQ Encoding
+
+   The vector to encode, X, is converted into an index i such that
+   0 <= i < V(N,K) as follows.  Let i = 0 and k = 0.  Then, for
+   j = (N - 1) down to 0, inclusive, do:
+
+   1.  If k > 0, set i = i + (V(N-j-1,k-1) + V(N-j,k-1))/2.
+
+   2.  Set k = k + abs(X[j]).
+
+   3.  If X[j] < 0, set i = i + (V(N-j-1,k) + V(N-j,k))/2.
+
+   The index i is then encoded using the procedure in Section 5.1.4 with
+   ft = V(N,K).
+
+6.  Conformance
+
+   It is our intention to allow the greatest possible choice of freedom
+   in implementing the specification.  For this reason, outside of the
+   exceptions noted in this section, conformance is defined through the
+   reference implementation of the decoder provided in Appendix A.
+   Although this document includes a prose description of the codec,
+   should the description contradict the source code of the reference
+   implementation, the latter shall take precedence.
+
+   Compliance with this specification means that, in addition to
+   following the normative keywords in this document, a decoder's output
+   MUST also be within the thresholds specified by the opus_compare.c
+   tool (included with the code) when compared to the reference
+   implementation for each of the test vectors provided (see
+   Appendix A.4) and for each output sampling rate and channel count
+   supported.  In addition, a compliant decoder implementation MUST have
+   the same final range decoder state as that of the reference decoder.
+   It is therefore RECOMMENDED that the decoder implement the same
+   functional behavior as the reference.  A decoder implementation is
+   not required to support all output sampling rates or all output
+   channel counts.
+
+6.1.  Testing
+
+   Using the reference code provided in Appendix A, a test vector can be
+   decoded with
+
+      opus_demo -d <rate> <channels> testvectorX.bit testX.out
+
+   where <rate> is the sampling rate and can be 8000, 12000, 16000,
+   24000, or 48000, and <channels> is 1 for mono or 2 for stereo.
+
+
+
+
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+
+
+   If the range decoder state is incorrect for one of the frames, the
+   decoder will exit with "Error: Range coder state mismatch between
+   encoder and decoder".  If the decoder succeeds, then the output can
+   be compared with the "reference" output with
+
+      opus_compare -s -r <rate> testvectorX.dec testX.out
+
+   for stereo or
+
+      opus_compare -r <rate> testvectorX.dec testX.out
+
+   for mono.
+
+   In addition to indicating whether the test vector comparison passes,
+   the opus_compare tool outputs an "Opus quality metric" that indicates
+   how well the tested decoder matches the reference implementation.  A
+   quality of 0 corresponds to the passing threshold, while a quality of
+   100 is the highest possible value and means that the output of the
+   tested decoder is identical to the reference implementation.  The
+   passing threshold (quality 0) was calibrated in such a way that it
+   corresponds to additive white noise with a 48 dB SNR (similar to what
+   can be obtained on a cassette deck).  It is still possible for an
+   implementation to sound very good with such a low quality measure
+   (e.g., if the deviation is due to inaudible phase distortion), but
+   unless this is verified by listening tests, it is RECOMMENDED that
+   implementations achieve a quality above 90 for 48 kHz decoding.  For
+   other sampling rates, it is normal for the quality metric to be lower
+   (typically, as low as 50 even for a good implementation) because of
+   harmless mismatch with the delay and phase of the internal sampling
+   rate conversion.
+
+   On POSIX environments, the run_vectors.sh script can be used to
+   verify all test vectors.  This can be done with
+
+      run_vectors.sh <exec path> <vector path> <rate>
+
+   where <exec path> is the directory where the opus_demo and
+   opus_compare executables are built and <vector path> is the directory
+   containing the test vectors.
+
+6.2.  Opus Custom
+
+   Opus Custom is an OPTIONAL part of the specification that is defined
+   to handle special sample rates and frame rates that are not supported
+   by the main Opus specification.  Use of Opus Custom is discouraged
+   for all but very special applications for which a frame size
+   different from 2.5, 5, 10, or 20 ms is needed (for either complexity
+   or latency reasons).  Because Opus Custom is optional, streams
+
+
+
+Valin, et al.                Standards Track                  [Page 156]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   encoded using Opus Custom cannot be expected to be decodable by all
+   Opus implementations.  Also, because no in-band mechanism exists for
+   specifying the sampling rate and frame size of Opus Custom streams,
+   out-of-band signaling is required.  In Opus Custom operation, only
+   the CELT layer is available, using the opus_custom_* function calls
+   in opus_custom.h.
+
+7.  Security Considerations
+
+   Like any other audio codec, Opus should not be used with insecure
+   ciphers or cipher-modes that are vulnerable to known plaintext
+   attacks.  In addition to the zeros used in Opus padding, digital
+   silence frames generate predictable compressed results and the TOC
+   byte may have an easily predictable value.
+
+   Implementations of the Opus codec need to take appropriate security
+   considerations into account, as outlined in [DOS].  It is extremely
+   important for the decoder to be robust against malicious payloads.
+   Malicious payloads must not cause the decoder to overrun its
+   allocated memory or to take an excessive amount of resources to
+   decode.  Although problems in encoders are typically rarer, the same
+   applies to the encoder.  Malicious audio streams must not cause the
+   encoder to misbehave because this would allow an attacker to attack
+   transcoding gateways.
+
+   The reference implementation contains no known buffer overflow or
+   cases where a specially crafted packet or audio segment could cause a
+   significant increase in CPU load.  However, on certain CPU
+   architectures where denormalized floating-point operations are much
+   slower than normal floating-point operations, it is possible for some
+   audio content (e.g., silence or near silence) to cause an increase in
+   CPU load.  Denormals can be introduced by reordering operations in
+   the compiler and depend on the target architecture, so it is
+   difficult to guarantee that an implementation avoids them.  For
+   architectures on which denormals are problematic, adding very small
+   floating-point offsets to the affected signals to prevent significant
+   numbers of denormalized operations is RECOMMENDED.  Alternatively, it
+   is often possible to configure the hardware to treat denormals as
+   zero (DAZ).  No such issue exists for the fixed-point reference
+   implementation.
+
+   The reference implementation was validated in the following
+   conditions:
+
+   1.  Sending the decoder valid packets generated by the reference
+       encoder and verifying that the decoder's final range coder state
+       matches that of the encoder.
+
+
+
+
+Valin, et al.                Standards Track                  [Page 157]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   2.  Sending the decoder packets generated by the reference encoder
+       and then subjected to random corruption.
+
+   3.  Sending the decoder random packets.
+
+   4.  Sending the decoder packets generated by a version of the
+       reference encoder modified to make random coding decisions
+       (internal fuzzing), including mode switching, and verifying that
+       the range coder final states match.
+
+   In all of the conditions above, both the encoder and the decoder were
+   run inside the Valgrind [VALGRIND] memory debugger, which tracks
+   reads and writes to invalid memory regions as well as the use of
+   uninitialized memory.  There were no errors reported on any of the
+   tested conditions.
+
+8.  Acknowledgements
+
+   Thanks to all other developers, including Henrik Astrom, Jon
+   Bergenheim, Raymond Chen, Soren Skak Jensen, Gregory Maxwell,
+   Christopher Montgomery, and Karsten Vandborg Sorensen.  We would also
+   like to thank Igor Dyakonov, Hoang Thi Minh Nguyet, Christian Hoene,
+   Gian-Carlo Pascutto, and Jan Skoglund for their help with testing of
+   the Opus codec.  Thanks to Andrew D'Addesio, Elwyn Davies, Ralph
+   Giles, John Ridges, Ben Schwartz, Kat Walsh, Mark Warner, Keith Yan,
+   and many others on the Opus and CELT mailing lists for their bug
+   reports and feedback.  At last, the authors would like to thank
+   Robert Sparks, Cullen Jennings, and Jonathan Rosenberg for their
+   support throughout the standardization process.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 158]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+9.  References
+
+9.1.  Normative References
+
+   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
+              Requirement Levels", BCP 14, RFC 2119, March 1997.
+
+9.2.  Informative References
+
+   [BURG]     Burg, J., "Maximum Entropy Spectral Analysis", Proceedings
+              of the 37th Annual International SEG Meeting, Vol.
+              6, 1975.
+
+   [CELT]     Valin, JM., Terriberry, T., Maxwell, G., and C.
+              Montgomery, "Constrained-Energy Lapped Transform (CELT)
+              Codec", Work in Progress, July 2010.
+
+   [CODING-THESIS]
+              Pasco, R., "Source coding algorithms for fast data
+              compression", Ph.D. thesis Dept. of Electrical
+              Engineering, Stanford University, May 1976.
+
+   [DOS]      Handley, M., Rescorla, E., and IAB, "Internet Denial-of-
+              Service Considerations", RFC 4732, December 2006.
+
+   [FFT]      Wikipedia, "Fast Fourier Transform",
+              <http://en.wikipedia.org/w/
+              index.php?title=Fast_Fourier_transform&oldid=508004516>.
+
+   [GOOGLE-NETEQ]
+              "Google NetEQ code", <http://code.google.com/p/webrtc/
+              source/browse/trunk/src/modules/audio_coding/NetEQ/main/
+              source/?r=583>.
+
+   [GOOGLE-WEBRTC]
+              "Google WebRTC code", <http://code.google.com/p/webrtc/>.
+
+   [HADAMARD] Wikipedia, "Hadamard Transform", <http://en.wikipedia.org/
+              w/index.php?title=Hadamard_transform&oldid=508252957>.
+
+   [KABAL86]  Kabal, P. and R. Ramachandran, "The Computation of Line
+              Spectral Frequencies Using Chebyshev Polynomials", IEEE
+              Trans. Acoustics, Speech, Signal Processing, Vol. 34, no.
+              6, pp. 1419-1426, December 1986.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 159]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   [LAROIA-ICASSP]
+              Laroia, R., Phamdo, N., and N. Farvardin, "Robust and
+              Efficient Quantization of Speech LSP Parameters Using
+              Structured Vector Quantization", ICASSP-1991, Proc. IEEE
+              Int. Conf. Acoust., Speech, Signal Processing, pp. 641-
+              644, October 1991.
+
+   [LPC]      Wikipedia, "Linear Prediction", <http://en.wikipedia.org/
+              w/index.php?title=Linear_prediction&oldid=497201278>.
+
+   [MARTIN79] Martin, G., "Range encoding: An algorithm for removing
+              redundancy from a digitised message", Proc. Institution of
+              Electronic and Radio Engineers International Conference on
+              Video and Data Recording, 1979.
+
+   [MATROSKA-WEBSITE]
+              "Matroska website", <http://matroska.org/>.
+
+   [MDCT]     Wikipedia, "Modified Discrete Cosine Transform", <http://
+              en.wikipedia.org/w/
+              index.php?title=Modified_discrete_cosine_
+              transform&oldid=490295438>.
+
+   [OPUS-GIT] "Opus Git Repository", <https://git.xiph.org/opus.git>.
+
+   [OPUS-WEBSITE]
+              "Opus website", <http://opus-codec.org/>.
+
+   [PRINCEN86]
+              Princen, J. and A. Bradley, "Analysis/Synthesis Filter
+              Bank Design Based on Time Domain Aliasing Cancellation",
+              IEEE Trans. Acoustics, Speech, and Siginal Processing,
+              ASSP-34 (5), pp. 1153-1161, October, 1986.
+
+   [PVQ]      Fischer, T., "A Pyramid Vector Quantizer", IEEE Trans. on
+              Information Theory, Vol. 32, pp. 568-583, July 1986.
+
+   [RANGE-CODING]
+              Wikipedia, "Range Coding", <http://en.wikipedia.org/w/
+              index.php?title=Range_encoding&oldid=509582757>.
+
+   [REQUIREMENTS]
+              Valin, JM. and K. Vos, "Requirements for an Internet Audio
+              Codec", RFC 6366, August 2011.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 160]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   [RFC3533]  Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
+              RFC 3533, May 2003.
+
+   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
+              Jacobson, "RTP: A Transport Protocol for Real-Time
+              Applications", STD 64, RFC 3550, July 2003.
+
+   [SCHUR]    Le Roux, J. and C. Gueguen, "A fixed point computation of
+              partial correlation coefficients", ICASSP-1977, Proc. IEEE
+              Int. Conf. Acoustics, Speech, and Signal Processing, pp.
+              257-259, June 1977.
+
+   [SILK]     Vos, K., Jensen, S., and K. Sorensen, "SILK Speech Codec",
+              Work in Progress, September 2010.
+
+   [SPECTRAL-PAIRS]
+              Wikipedia, "Line Spectral Pairs", <http://
+              en.wikipedia.org/w/
+              index.php?title=Line_spectral_pairs&oldid=365426016>.
+
+   [SRTP-VBR] Perkins, C. and JM. Valin, "Guidelines for the Use of
+              Variable Bit Rate Audio with Secure RTP", RFC 6562,
+              March 2012.
+
+   [VALGRIND] "Valgrind website", <http://valgrind.org/>.
+
+   [VALIN2010]
+              Valin, JM., Terriberry, T., Montgomery, C., and G.
+              Maxwell, "A High-Quality Speech and Audio Codec With Less
+              Than 10 ms Delay", IEEE Trans. on Audio, Speech, and
+              Language Processing, Vol. 18, No. 1, pp. 58-67 2010.
+
+   [VECTORS-PROC]
+              "Opus Testvectors (proceedings)", <http://www.ietf.org/
+              proceedings/83/slides/slides-83-codec-0.gz>.
+
+   [VECTORS-WEBSITE]
+              "Opus Testvectors (website)",
+              <http://opus-codec.org/testvectors/>.
+
+   [VITERBI]  Wikipedia, "Viterbi Algorithm", <http://en.wikipedia.org/
+              w/index.php?title=Viterbi_algorithm&oldid=508835871>.
+
+   [VORBIS-WEBSITE]
+              "Vorbis website", <http://xiph.org/vorbis/>.
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 161]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   [WHITENING]
+              Wikipedia, "White Noise", <http://en.wikipedia.org/w/
+              index.php?title=White_noise&oldid=497791998>.
+
+   [Z-TRANSFORM]
+              Wikipedia, "Z-transform", <http://en.wikipedia.org/w/
+              index.php?title=Z-transform&oldid=508392884>.
+
+   [ZWICKER61]
+              Zwicker, E., "Subdivision of the Audible Frequency Range
+              into Critical Bands", The Journal of the Acoustical
+              Society of America, Vol. 33, No 2 pp. 248, February 1961.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 162]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+Appendix A.  Reference Implementation
+
+   This appendix contains the complete source code for the reference
+   implementation of the Opus codec written in C.  By default, this
+   implementation relies on floating-point arithmetic, but it can be
+   compiled to use only fixed-point arithmetic by defining the
+   FIXED_POINT macro.  The normative behavior is defined as the output
+   using the floating-point configuration.  Information on building and
+   using the reference implementation is available in the README file.
+
+   The implementation can be compiled with either a C89 or a C99
+   compiler.  It is reasonably optimized for most platforms such that
+   only architecture-specific optimizations are likely to be useful.
+   The Fast Fourier Transform (FFT) [FFT] used is a slightly modified
+   version of the KISS-FFT library, but it is easy to substitute any
+   other FFT library.
+
+   While the reference implementation does not rely on any _undefined
+   behavior_ as defined by C89 or C99, it relies on common
+   _implementation-defined behavior_ for two's complement architectures:
+
+   o  Right shifts of negative values are consistent with two's
+      complement arithmetic, so that a>>b is equivalent to
+      floor(a/(2**b)),
+
+   o  For conversion to a signed integer of N bits, the value is reduced
+      modulo 2**N to be within range of the type,
+
+   o  The result of integer division of a negative value is truncated
+      towards zero, and
+
+   o  The compiler provides a 64-bit integer type (a C99 requirement
+      which is supported by most C89 compilers).
+
+   In its current form, the reference implementation also requires the
+   following architectural characteristics to obtain acceptable
+   performance:
+
+   o  Two's complement arithmetic,
+
+   o  At least a 16 bit by 16 bit integer multiplier (32-bit result),
+      and
+
+   o  At least a 32-bit adder/accumulator.
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 163]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+A.1.  Extracting the Source
+
+   The complete source code can be extracted from this document, by
+   running the following command line:
+
+   o  cat rfc6716.txt | grep '^\ \ \ ###' | sed -e 's/...###//' | base64
+      --decode > opus-rfc6716.tar.gz
+
+   o  tar xzvf opus-rfc6716.tar.gz
+
+   o  cd opus-rfc6716
+
+   o  make
+
+   On systems where the provided Makefile does not work, the following
+   command line may be used to compile the source code:
+
+   o  cc -O2 -g -o opus_demo src/opus_demo.c `cat *.mk | grep -v fixed |
+      sed -e 's/.*=//' -e 's/\\\\//'` -DOPUS_BUILD -Iinclude -Icelt
+      -Isilk -Isilk/float -DUSE_ALLOCA -Drestrict= -lm
+
+   On systems where the base64 utility is not present, the following
+   commands can be used instead:
+
+   o  cat rfc6716.txt | grep '^\ \ \ ###' | sed -e 's/...###//' >
+      opus.b64
+
+   o  openssl base64 -d -in opus.b64 > opus-rfc6716.tar.gz
+
+   The SHA1 hash of the opus-rfc6716.tar.gz file is
+   86a927223e73d2476646a1b933fcd3fffb6ecc8c.
+
+A.2.  Up-to-Date Implementation
+
+   As of the time of publication of this memo, an up-to-date
+   implementation conforming to this RFC is available in a Git
+   repository [OPUS-GIT].  Releases and other resources are available at
+   [OPUS-WEBSITE].  However, although that implementation is expected to
+   remain conformant with the RFC, it is the code in this document that
+   shall remain normative.
+
+A.3.  Base64-Encoded Source Code
+
+   ###H4sIAEeqNVACA+xde3PixrLfv/kUfXarEvDF2ODHZrNJajHIWCe8LgJ7t+65pSOL
+   ###AXRWSEQPe52c891v98xISEI8vMZ7kyqTikHSdE93T0/Pr+ehdRehf+hNzPO31fOj
+   ###V8/zOcbP27dn/Bs/2W/+u1o7rp6d1M6qx29fHVfxv7NXcPbqL/TJKvcX+bjJ9m/0
+   ###+p/Ubus5THN+frqu/U9O3lZl+x/XzslPqugF1Vfw6qX9n/3TcBcPnjWdBVB99+70
+   ###EFuhCqoyvIShF/pBGT5ai1ml503LoH1+WDBoW3MrYOMy9MzA8C2zXICcz9+Z4Rx2
+
+
+
+Valin, et al.                Standards Track                  [Page 164]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###DM+Ea8O2nDIMrbkbzB7gogJD5nnWLf55yKdtaOqgV4aWx6au9wAd48s9s+0y/vA+
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+   ###YpmcYxkMj8GCeWh1km7huXfWGH9waYIZQya27d6jQcB0HSGUXyCiOQt+LBQOIS2R
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+   ###aghXvXZTGWhQ7zbxbnc4UC9Gw95AK/zzn3UNSb//nj+qdz+B8rE/UDQNegNQO/22
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+   ###4dINnbGA5PlJAADcCJQDtw/ZxAEOjgpHBwUqtK88ATJ5At3YQ5oAe0kTIJEm0MWT
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+   ###vuWwvEdIJrAUvCaBKrPXnBMx0nVzYYc+/V/AcQ+DE7xuvIY/Cm8wlFiTQiF4WDAq
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+   ###oViigdRjHOxxMj5QCj5wP0MoiXax/BniS+qTXG8csiggpXvHwdqqTLwKWFFuGuDF
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+   ###LcixssKfRKqW3ou+oCbcH4GqjWKPH/Bb9JDx+h6x4rRU/VpPpV4W+WnGuwmppuRM
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+   ###Axw2RYEwx2Mp59/RbYXDrnfXTQ6L6ZUfxE5bPYeDhTlPtp6a9CWDXAzbDgcLyqt8
+   ###cmbmV6AtVKVJBQ8zON3HfnoQN4WsijRYPt7o0JIx5WCCZKWnSJ4ZLyEjp3tpLwyI
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+   ###5bpU0p+suP+iKsi3Eum92bE2udbXOtYm13oux9rkWvtyLIDItVos8LnMZDIyA0qX
+   ###bm6R14Qeo4lFTOM86YxEQHfSPpUY+NYOr1MW8BYqft2QJEJ+blTPjD/wuPEn0eOa
+
+
+
+Valin, et al.                Standards Track                  [Page 165]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###zIhhN0cVuTA7o/uda423I+2VzpRoCuq7Pn0t5NSksxHmPwbTmIFdzOnHROOx30JG
+   ###F5VKJRJmt6xjzB6Xdcjy+8s6ojz7YG1VT8g6UDbBBTCCPRWULZl9LRRL+70EecnW
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+
+
+
+Valin, et al.                Standards Track                  [Page 166]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###J35A8zFbwbccDHHAWlsB594xvljzcB6fAM9hzI15QTh3wpMO/HHtqv0jcjSX9tnI
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+
+
+
+Valin, et al.                Standards Track                  [Page 167]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###xzIl1mZxwqvNIy26m7aafIDsZpLcxeCUpMa678WIsYTG4uiRMenXC056MQRR1+1m
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+
+
+
+Valin, et al.                Standards Track                  [Page 168]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###w3yV2AZ92FwWuWaspAotMVIz0T1zUqjcuWvx5weR1Nma9rf5Qz/cNlqWuKNHR1fZ
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+
+
+
+Valin, et al.                Standards Track                  [Page 169]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###kUEJj1hhNCBO/fRqCs4HpBsb6od9/8rJVVbdPC6d1HcIM717aV0z265GBIjlCYje
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+
+
+
+Valin, et al.                Standards Track                  [Page 170]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###jJaTgQ7kLIW2xMlJq0z5oiuD9pghqn9BL7JOMMY0HA7rs6WjEEUV59DLFFwwR7kz
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+
+
+
+Valin, et al.                Standards Track                  [Page 171]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###ORSVGLrQsgwZHQYieTomeZIw9ru/AVuMXHkJWo2C3hTN7ilpOkaUZvOBT0EEB+Wt
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+
+
+
+Valin, et al.                Standards Track                  [Page 172]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###zZmj5rVMoY0xzIcQB03qhB+MuEtyG+ouKYt10xZ74s5BMnBfj1Uzu5zLsSmYpDd0
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+
+
+
+Valin, et al.                Standards Track                  [Page 173]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###24cDfAxybQiSsrwAFO6Bqr1mrWRc0IvcwZTB+1MQiZEheQb6VVmpNLx6zatVaRDo
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+
+
+
+Valin, et al.                Standards Track                  [Page 174]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###h0G6kITejqIbJrJE3AOYBamE4NDPYgboLI1L4jJHENoVjQlCe3Ib9Ptl6eNLtLBZ
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+
+
+
+Valin, et al.                Standards Track                  [Page 175]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###k3NK9hM3y2SR3AqJKN0jfYJdUIzz/x0L+v/1gOB9PSB4CwDBfDXOsLtKP7qnM3/c
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+
+
+
+Valin, et al.                Standards Track                  [Page 176]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###639qa9VaI63/qS71P3+O/mcR9c+JH904uyNgZCKOK5FIwvLiUXogkU5mRh6YFy+e
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+
+
+
+Valin, et al.                Standards Track                  [Page 177]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###55S9wjh0VrEDkJsKTnL41ui5mh6oDI66Wnjh4P/o5Dzq+Jj7E72jhkEniGPKJiOy
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+
+
+
+Valin, et al.                Standards Track                  [Page 178]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###dZGx6BGB4CpyEWiWVs85yil4TYtOao/RUBRjHogYM+OAPk10JmcLhV9qYLxccX4R
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+
+
+
+Valin, et al.                Standards Track                  [Page 179]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###X+zs/SgYICNyY+7H7UoJhDsF7bwwR/eRhfVRwTUMeIgidKNV+3TbnV97cjsCDhjY
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+
+
+
+Valin, et al.                Standards Track                  [Page 180]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 181]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 182]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###L9kCg+N8N5ti/MCY6NRafv/Wv4/ZTkZkbyOTRrrbHmHGObwyRY4+nW3DiKEhb682
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+
+
+
+Valin, et al.                Standards Track                  [Page 183]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 184]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###R6xAElYOQEp8sct3xqmoD6m4kdadWUpR/4To25nxpjkeVuKBNytulhHxKuNtQgmh
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+
+
+
+Valin, et al.                Standards Track                  [Page 185]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###kDm7IWc66IxAZAzhsB1O4ldOZWUDBL2VdfzVqOPvOq5JBl7PWiZsuoWNfqjRQnkb
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+
+
+
+Valin, et al.                Standards Track                  [Page 186]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+   ###5UHg58v/bq1R9xLyf63uLeX/pfy/lP+X8v9S/l/K/0v5fyn/f9X47wbP80AA9614
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+   ###8PnTh0tDLA6iiKUW4Q56CEdoCdjZ6QQ/0avDI0B25CXheQ9esJstzHSPJio4ZPjd
+
+
+
+Valin, et al.                Standards Track                  [Page 187]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###VNlE9Vv2wYmEk6X1is2RIrL40MJ5MEza0FKMxxYHZsacU0OQ+0oUF1k70aP3WQvo
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+   ###Ter/apX6Uv/3V+n/3LT+7yIcjCbX987uinMRRFHYhl/3c1V/C7sKLH0Blvq2pb5t
+   ###qW9b6tuW+rbn1Lepe2fLISD/4ltntkuAoVWLW/F0jMlvH3AIyFSzDXrR8AqoN6nV
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+   ###bZROTn/ZWP9MpOA/j04RqQ4OcCnOxXs6bqZAVKLmS3gyQWOz1yun8B+b9E7CST9o
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+   ###59bbo9GN6OFnINUj57++c/ZB1oDDpjOK6NZtD5aA58xV7gM/arobcujdLkp6zZfn
+
+
+
+Valin, et al.                Standards Track                  [Page 188]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###QBWvQSSajIQt8gAI8HXz/0zxtEfriv8kcnF88DtBxJr3Dh5Efhg5J6NeDyNkA7TP
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+   ###YnnuD6tY1hKzht3yUTDt/umjF8XKxWIu0VJ+0SmtFt6QmV2MqQopCWFJBYLwgR1H
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+   ###0POY2CzAmLOTsO+EE6g5RHYH6BEyIzjs2+uAuCygFGjYjuSjM5lCi9wJhi0g86NO
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+   ###lXkwxHi3sb4/l4o5seKMSJPraez0/IiFxbbUjGEE5CgKOpP+PSD3lR+Rkgw7IMWi
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+
+
+
+Valin, et al.                Standards Track                  [Page 189]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###RkKZKPx5ker7FKS3ROp+ZzpeMaIJ5B66sUmdEdrPX/uf95vlvvYsF8j2PaEahlMu
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+   ###7sMSHsBZuAajLMtSkV3qs87QB8NI6DpKFDSiTE73MSrpa6unYhiSqVT8JE+r5GiW
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+   ###3zBfATRP2IOpvUc7itGsh5stjfQKWXu4Q63tktxUdrlkO+VEP+Hz7SuXt1V57rby
+   ###eFcVk7tq08zThjnJTMKLbeOjAqOB9cbDRGTJN5o2nx2AuHWmRIaUrNj2Ozd/lhRl
+   ###MH1/G3lq9nvvzxa4RAnv310mU7xA8DcSoB4vM2WS3AeZGe9P5WayKalgK5CBLDkW
+   ###ZyHJDvKLD7MnyYZm0VLRWvj8LEp5IRalOINFUVxumm4ys5tNNQUaHw2BMMGAE6zN
+
+
+
+Valin, et al.                Standards Track                  [Page 190]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 191]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###49IaPOnbw9nNdGw0Mtq2I1JD776Lmz4Dl9aK5d4iG4o+iyzlEpJ6qyRQ8uRQZA5U
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+
+
+
+Valin, et al.                Standards Track                  [Page 192]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###3H1zsrcPeoyv34CO85ce6GXf7wK0tj+wRRC0bLh6CFhVGQMmCkOkKmEs8Ef+G+gu
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+
+
+
+Valin, et al.                Standards Track                  [Page 193]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###7ykR2J2VKvU+P6cAWWlUn/PTOipNK8q6+mk9H046lYz5207rF8+qjYlHz5xrzxyM
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+
+
+
+Valin, et al.                Standards Track                  [Page 194]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###n3Nvqxe5WNOH0JHHCDd0v6QhmKy3IGDOp7dnsp/KKOKkF2DQmLBv/gCMPrBvN/+1
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+
+
+
+Valin, et al.                Standards Track                  [Page 195]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###3jjntzdjEmUkn86JRzU5cTSU+zQESY4Skryo2BOCMfw4nN1/QphbCmjMbn05BQZ1
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+
+
+
+Valin, et al.                Standards Track                  [Page 196]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###SBghDJ6eAhxDGEAfUCQIU7wULYj4ZZRhdRgulYyTDIskHgIRwPzTr4SKJFlIJOQh
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+
+
+
+Valin, et al.                Standards Track                  [Page 197]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###jNLd5los3ukNiHsuE9OWmOF9LOBt9sgSXjiojB1FvM1kWcbbfJiFvLBqwTNKeeHm
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+
+
+
+Valin, et al.                Standards Track                  [Page 198]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###UsBlImdVOE6VTi1kHPbHAZ1kU2ITicS6NUKP3J0Zh64HZERj+YjKhxLVF9DwcBwE
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+
+
+
+Valin, et al.                Standards Track                  [Page 199]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###JYvBlA2QUeRaizNyl8Pi8iTCgxywtIMY835QTNExEtTkxxw7EmP4FzqzyRXAAVAg
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+
+
+
+Valin, et al.                Standards Track                  [Page 200]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###7xpZtuHZ0sOkiT23rRCbTSiHWP8SUFgc8TmI7Px4cXSy1fq0K/NP3v8BTrk6+Eft
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+   ###xbmb8wIRQ3db3h+r6l9B/XHrhUWjVLeOkpDnDIsReBwLAorys+lk+AwZkh2vh+Qp
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+
+
+
+Valin, et al.                Standards Track                  [Page 201]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###iqkBd9Muql7n+CF0vIWkteKr+TMyg/ElfTsFZscf4X/A1HURaNuCpDXf2IC2/244
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+
+
+
+Valin, et al.                Standards Track                  [Page 202]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###KPghicIkS6SBtBv5kc/94BwFXuRRAyFozDCHXCcMEtCYM3mIAs/rhtxpGCN+UkZl
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+
+
+
+Valin, et al.                Standards Track                  [Page 203]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###rPlQpiiVP2YpLmVFQ5kHpg5QoHxELhAliiekSH6hIs3EA8T6moagFfQEC4EBZF43
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+
+
+
+Valin, et al.                Standards Track                  [Page 204]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###jmEnkvggiBPJBkLn0zmTCh3cTDoNOpiKh+dDLB+FKnkAnaAPKZkGZbCQdAp0xD1M
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+
+
+
+Valin, et al.                Standards Track                  [Page 205]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###ApCk54dqmUUp3s4iENj73bD+slTdIUNhIomia7p/QaFK759TA2ApR2rYAGeo77QI
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+
+
+
+Valin, et al.                Standards Track                  [Page 206]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###0Wond81mJ3cH2Wdraui97Mg14Hl/hHtpF4glZwFf+h/7o3H/bMyUSAhT+3GnR6fb
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+
+
+
+Valin, et al.                Standards Track                  [Page 207]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###PJf3TRMMMg/TJMQYFe3rwf0qjH6A/4EYirsxHoDETKlZHOMGIm6ewjs/E/657DTE
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+
+
+
+Valin, et al.                Standards Track                  [Page 208]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###xgqdq4JyfgNG7jnDGRGYdE9pqAGpHWL/Zj5ccOXw/eavQcfvUJ2qCahrZ3oEFsgp
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+
+
+
+Valin, et al.                Standards Track                  [Page 209]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###W23SSjyD92cbL9fPcmz+19xaWuO8brPxsP9RH4wxlPjpCkP0zqfqE1KscrWa9fjA
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+
+
+
+Valin, et al.                Standards Track                  [Page 210]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###EItMA4KmoiJOMwgEhAdhSzIBV4jqdmOT+6ngdSZnoKQXnYD5YUjwf2QuJbuoWTVL
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+
+
+
+Valin, et al.                Standards Track                  [Page 211]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 212]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###nT7A7uIcuwja5szze9f8DVzcerKSdlUn21TNifIDf6/kml4pU8J7t0IltgGorGPD
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+
+
+
+Valin, et al.                Standards Track                  [Page 213]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###xV4hMlqHY7YaFYHJdhN+oUm/3GRlMGWezOykjLKxq3H9/tFhj2XRWdPtki5hTt9/
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+
+
+
+Valin, et al.                Standards Track                  [Page 214]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###ONSTxTp6MNACveb3DZfeccrke8i/9g8sLjIwGmCF8mroofqeCavDiz6R05/9WOVe
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+   ###hEvfyNtKbzHFr8p0w1rE1yoijDiwaux9FocMi5/hPUeaS9MrLdUtDeMtn8T3PoRb
+
+
+
+Valin, et al.                Standards Track                  [Page 215]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###iiRQJfa/zMYTJSuwXuxAdTnPnQ7RWx4TXmhEdEsmDhovnqr5HKMsmomlwSPCJypQ
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+   ###xWP1W+Ye7I86Tv4MhEu5m5KWUtK4q6Kl0tNvrWRkK/QWSFHoLZSX0JJaKDMvYSKC
+   ###/0J5CW0JjXLkJfS+ZTLCHCvuvHF58OYPtNpSjXz2bITiFknj+PBfNKXeHzdv4T95
+   ###sj/tTtAfKOdfolp21r9CgdMnV5MIhyhmkM0mypkmBGWR5EKtQK+xBkx9+OEZSv5y
+   ###zr4Lp52OH4bfSeb5XTAAxAZdBsOb3vmDSfxGRFGdDIcsvANCUuvAUsZrOwQpfo4B
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+   ###DJErOEhw/MdLVt4xdLrvpoNPA9jsRbdT0uDo2CvTz480LYlJwc0FT2ahX6JCEZON
+
+
+
+Valin, et al.                Standards Track                  [Page 216]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###iJmJwPaDayzOvuPz/a5+0WycnUZGsqu//x0YgLIGviv3pr/8Aq19l4wLhB350+rz
+   ###L/VB4iiPe53tHXf7KdoGnpKrzMbtEmFgevHt7U36Cx/jr1vZqmz/yfUq7la14m3v
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+   ###dwMbeC9Cm10/sP/ttwflk/a4w94BVx6Q47nRDs+8SrYZCtLWw1MU+OvfT8btDu4B
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+   ###P3ZkivffBiwcTscdn55cB4M29Ayrhg6P9zQc09/hdIKtgFgW9IKOiBXZHvsY1PMO
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+   ###4RbQFldz2tegk8ErMdHYBuyGATpuUjA5MndBGzFMGpveIYDY6beDO3+8kdILgKag
+   ###Q/YCBtmddvxv0RFx9YgOyIYdkiHacq6ewjQM4eWYwZyC5N/uhzG+ZVwupg5ADuvU
+   ###D6geBVjC0ErQIbr3O/AnrDnsBD66hNLyABjqMhnwsEw0LArqhfMy8jtIADgmDmo4
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+   ###/GmOFLogfXz22msdwYS0LpF6P3ibH/foiL9CNPXMcav4xd123OeO5zoe/fQ2HW/b
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+   ###/b4rnjo/lGDfntCm0YZGy6CGxF1HFeYGNxp0U8cd4AfaAYG3gkbd/+zjTgF7KOed
+   ###7hZpkBswiQIRpCBhpMZgUPVIwWsJtan1yeZ7orjv37XvTz+4W6rjCd6p2HGMP+7m
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+
+
+
+Valin, et al.                Standards Track                  [Page 217]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###QZLuZEwLKniOvT/AzoNiS3/22BMoQt+1cPzYSCdOpGcoxVFQf1NZllcNOjJLMbVH
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+   ###ce+Vrvygd8XobcY8ABuzNNfQm+Ol9CbhEQb9g07r9aOKWj+/JgIlifB1Sf6iRy1b
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+   ###Zl6Am4z5BTjsjdLsT6WPMffCQhSPKG7FGowHMQobbXA3vcMEiL5xQAu8Dkgdfn3x
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+   ###TX1lS0p0F2AspGJ/uH4blJQzF8SvarnAIaq/hT0OI3D+VJZhRFMOfiZ3I6GsumL/
+   ###AaiaWEP0Du/295X+mvEasBlMUUgrARv69VfqVqoBUPTZTZjuDoe4ZUWbRZz36ItP
+   ###Vpd2ZwIULqymxp5AiFkXHJxsxpg16KePam+NQ9LEEBQySOyGsn2TCi28TcaQwxr7
+   ###fPkrzRFxAKLtJ7xERfIttabN+NM1WKuDCbTOoy2MYPpCv0OOWTdo6fyP7n8OvnOg
+   ###GbomYxJIOnWotJeki/5wLZsUJBJfKmQAItmfVTr4R2EunPNe2NGdsKnywnH4KaOE
+   ###SXEL0xqHk0pjiKToSEFbsuvxo69KUpsA9id+LiAORYe0Dzjsy3BMeZwwuvGX9hhe
+   ###Cf80vPsSWZ9oPhVuiRO7G/8ulxNzS5fBKLXXrvlU3a55+L3IqvwT5SaKmlXxeOqD
+   ###rkadFuI/ph/E9rjLXpsff4jDXhCyfXQ7xdRTaF99UCZRZplSTBuNgdImBqJxUJ7s
+
+
+
+Valin, et al.                Standards Track                  [Page 218]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 219]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 220]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 221]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###umtciECryA6xEnC5z8Ogy7I3CL4vUMGMz1oT5DY96HPj6RnejxDN4inHRLd0Cy6r
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+
+
+
+Valin, et al.                Standards Track                  [Page 222]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###0iTHZ9C1LUGS65rc7W6vVxVENXqEAYmnax8jywKCMRikk0WASAtpHX6ZnCKVRtLH
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+   ###DvTVhTLH8+ORMO01MA0ZeiwtXy1eeQMgfhgKNgp/QMAHLXk4QtlWDiw1RexJUzOC
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+
+
+
+Valin, et al.                Standards Track                  [Page 223]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###VaC0LpsG1GNEBND9AtjLQLGUrt2PJwR+gMOsERgFIVgE2q9RzkL6Q2bTFIEYW1nl
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+
+
+
+Valin, et al.                Standards Track                  [Page 224]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###12glX8wNBQUYmcJ9+dLdLGHOOY6Ef0K7YVb+N0ya4LWqjzYDzor/4G26Zv63zS1v
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+   ###+l4SWyKwJb2AGcYrqHe2af5qEHnSyy728zM6jxPH49Bdh3yOSkDN21tb1e00qq2R
+
+
+
+Valin, et al.                Standards Track                  [Page 225]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###lI1LpktywgMnDwtWQALDEUOjZZyDZMeB5k29l8S7Z1Ah1XRb0m9ZFhFMCv1T1Dfo
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+   ###sWduRZjdMSJ0xd2SSm8C0CztPBEbf4banU/XzMcTKqn8YEF9VVNALKq5Za0srqan
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+   ###POSLu42/XbeCJMM8z6tsiYaqoiEmYsM8d3lDO9wy7j4Tzz3e4DPe4Ha1in/d58+9
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+
+
+
+Valin, et al.                Standards Track                  [Page 226]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###redU3916Rn+9bU4bVW/HXRgTm/NjwvO2OH/fqjyX5M74yMruplikFfFbjJAJ8vfE
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+
+
+
+Valin, et al.                Standards Track                  [Page 227]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###3YTOFSWtxd1w0u582mCvg89+yMI7TJU18sfkcAdjYWrCHU27qcnuze7yLAtbxJSO
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+
+
+
+Valin, et al.                Standards Track                  [Page 228]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###20dxFoOguRXikPuUvFMRqltJHwX3zqObnzNGceGX+aVhgBDfncX2K3lGIe6dbGWN
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+
+
+
+Valin, et al.                Standards Track                  [Page 229]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###Jhz5fufWSB8OxQ7rr65et5qXZxf1Fl4FxdConY1R504R7OwC08d4M9Jis6flSzWP
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+
+
+
+Valin, et al.                Standards Track                  [Page 230]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 231]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###9p+V/Wdl/1nZf1b2n5X9Z2X/Wdl/FrL/DNAARGJus/EaJwrFGrQCcQuO7ZU0G7FW
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+
+
+
+Valin, et al.                Standards Track                  [Page 232]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###JnHG5Med4VsU3GU2k+q3byhporN8g6sK+pmlCF5HHE7HGvjIO2x5oG1FeOQbjB0S
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+
+
+
+Valin, et al.                Standards Track                  [Page 233]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###FwxXe0jcN7mmX71KhVe0Q2OOFZckAHHYpcT61orwfcBkMIeNdwDH3RYdUZmLZfqo
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+
+
+
+Valin, et al.                Standards Track                  [Page 234]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###GZHJrDukaCNj7N3dcOIXOKqAlLvQ7c+SYAk54bA3+YJTLxNwR0BG4wAJbIxENCgQ
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+
+
+
+Valin, et al.                Standards Track                  [Page 235]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###yZPdwkgFms/r3Ie7dVw/fX35pnXSXDsKW5/e/DIXctf4gIeGOw5Pa4yvGDrIiGiz
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+
+
+
+Valin, et al.                Standards Track                  [Page 236]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 237]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###2uJ0mdBo+WF41Y5fK99VcFdJw50db5VMvMHXysZ2z3Ly5c5LNFkznTqS/DOfOrqs
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+   ###swnlV/aflf1nZf9Z2X9W9p+V/Wdl/1nZf35T/1/TWGK62EbiUTHdjZY7A4V5vGjn
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+
+
+
+Valin, et al.                Standards Track                  [Page 238]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###oJA80otB/uK6lxIo/VOcHj29fWsSiNf+JIUstQu0hCyMM0ddEEiu0PVNmclbZkc6
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+
+
+
+Valin, et al.                Standards Track                  [Page 239]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###U61iyMhjZRW5ysZSaTxx7QEIcMtcZYTCXDl3EcKH/CUYZQ8V/gomlprKOf0u472Z
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+
+
+
+Valin, et al.                Standards Track                  [Page 240]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###aBaBrlsp07td3HvKKsfBOdZaSgOB2D1GZE/cIIB3j0LTmUpuNnGf42muPLcHRp7p
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+
+
+
+Valin, et al.                Standards Track                  [Page 241]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###OJqiOE/X7kVLP3jwpmFkJ9kox6GRMiEkI9mLxDoZhU7VEhpK3tmq5B2m755q2S7m
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+
+
+
+Valin, et al.                Standards Track                  [Page 242]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 243]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 244]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###Tceej4+Lnz+cygPPaFnwigWvaHoPw9xT0jqm7pB7rMZgsTSeTup6zOMxljHDO7jD
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+
+
+
+Valin, et al.                Standards Track                  [Page 245]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 246]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###tCgGKZuB2CqGWqReruGGOFZPoJR50CEvzqlwDwucF14yhF/n+AMI3+wZKWmfLAsS
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+
+
+
+Valin, et al.                Standards Track                  [Page 247]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###1aR178hnllj1V+qwrDM775QRllMQkNK+CJUAxvpFo1Y1Wei5dxxNm2/DUfsn7ldJ
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+
+
+
+Valin, et al.                Standards Track                  [Page 248]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###kRkMyuZdkfM8JaTruWNM2wSXHN2zgDz7bhDwBHQUG28Q/dUVLfEwp6pKkZZEjG8u
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+
+
+
+Valin, et al.                Standards Track                  [Page 249]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###KunSQk/0Jnqi7zCPdK18jaDcEcJzeESu5jfOoG/MuRptyS1FxFCBzp2Nn39jn0dg
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+
+
+
+Valin, et al.                Standards Track                  [Page 250]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###3AbPGs1f3oR0DBnQxCQN22qyTYl8eMs1lqshQn8G4qtAXBeWswKtk5L6Uxminm+I
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+
+
+
+Valin, et al.                Standards Track                  [Page 251]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###po7kJc+Xb17OqWO6Fb/DBpAkMnS2Z17VmI8cA47C2SiW1BSqxZa8o8lk0S1lgmrX
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+
+
+
+Valin, et al.                Standards Track                  [Page 252]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###wZhHD1upkzGRxPNLBoVVnZ4e7bnKhmM5PqOs49JygcQWlTpFaOes5FjIkmKqidhG
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+
+
+
+Valin, et al.                Standards Track                  [Page 253]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###h23GhGZswcZjekY75Yt6kLEtWJrvGLd21tN0Qaznv1LDhCUB1EviQczEZLF15VlT
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+
+
+
+Valin, et al.                Standards Track                  [Page 254]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###vwgG/E+hQ4waROonlUMyq8SYzLtsvW+dH3P9DJAs+6J12kEZVyh45Clx/EhbHBfp
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+
+
+
+Valin, et al.                Standards Track                  [Page 255]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###DWix+mbFv3nlP8wEiJa2K2/g/e4+Rym4dPmvvrq62ozkv+YGyH/NjbVGIf8V8l8h
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+
+
+
+Valin, et al.                Standards Track                  [Page 256]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###I8kdlJUc68TTxNYvv1TS6x/RtAgZkVibQFLBeqUjJPZDr6dmGskzbSPftOwszPMG
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+
+
+
+Valin, et al.                Standards Track                  [Page 257]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###k+GZsaEF2nGK57N4Povns3g+i+ezeD7/I5/PmBcXep791tqPyaMrzB7iTT0qpkoh
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+
+
+
+Valin, et al.                Standards Track                  [Page 258]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###/te/Yl8OnQd1SVGjd1Ejz9caaUA6dm5dTEPlRnhP6A5vHXAF1g3wB/QaOz4ycFKD
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+
+
+
+Valin, et al.                Standards Track                  [Page 259]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###GbncXJ2rddbbohABpJsOehQAie5XyV8ACSZy3TnXZuk1RKBNxaqv8Qoh5Vc5lxmO
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+
+
+
+Valin, et al.                Standards Track                  [Page 260]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###cTxP7hOSsaNf39unrU7HPjo5/XhhpZLWNDU8LOK3kYfm1e7Uu/Omj7Drafg0p/cV
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+
+
+
+Valin, et al.                Standards Track                  [Page 261]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###7HEC+IslPBFTIGKJLL3h4ICAHmDGa77S6HO6uhUV4OIe1kFgwIy94VYqWNuGA1o6
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+   ###NBjvdLS5ZYtMybFB1fbWzo42jbG9oEMExGiGejIIBRhrKtngwTf0JudRt/+J9L88
+
+
+
+Valin, et al.                Standards Track                  [Page 262]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###fblrS2lNg4W0wOn639paY62p6X83NuDrQv9b6H8L/W+h/y30v4X+t9D/Fvrfn6n/
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+   ###GG/moPRuX93bwT3yQ5W8N5f6WawTEj5n1vNGwFz5vXuvl12ucZ4ldm8c33cHJ1eV
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+   ###BVN67a1r5P955jnlAlF/G/u7tpj/R90lZahFsSD5FslLzHeX5CWKu/SUReZE1CWS
+
+
+
+Valin, et al.                Standards Track                  [Page 263]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###srEuTr7rjqZXUURHBCWhmB7izwssMWc9mpC9liM6LtVixsvWwGEYZ32Z6y5Zrdl0
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+
+
+
+Valin, et al.                Standards Track                  [Page 264]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###LrHPMrzfww7gdZT6sEnknh4SfXhKJ0Be6fWIwFSh91xeSfAmp0aGyrV0YfABFqCR
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+
+
+
+Valin, et al.                Standards Track                  [Page 265]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###B/x9/5fT1nFb9l1esSWnT7n3Cbo1RMPCKmsjHDa41cUzsZPZubitE4dTbaZYXs5U
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+
+
+
+Valin, et al.                Standards Track                  [Page 266]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###/P/L/P/L/P/L/P/L/P/L/P9/dv3HgyiZiJB/4cG8ytJ/3E3xggoV8qjTmaKd5FV2
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+
+
+
+Valin, et al.                Standards Track                  [Page 267]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###okAhShxFvlxAiwqfmfhkHgkq4DmVcbQR3tq1cDiC7fRn2ourWxkyxdlGKqq4uiQv
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+
+
+
+Valin, et al.                Standards Track                  [Page 268]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 269]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###4d/sdxCtTPQIL3tREp4FVYycv7oCyvlDT46gDAMst3N5G7KjruzRZ9iXVtfsDhxH
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+
+
+
+Valin, et al.                Standards Track                  [Page 270]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 271]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###X12vYCtn6Qa3IkojryhUAQWCdYB24qUqsVQllqpEW7X+PUGT+H6KAmfh6YlrSFUF
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+
+
+
+Valin, et al.                Standards Track                  [Page 272]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###zfWKgktDCRs/7Y70XKoitxUiDx85+5i15FztdK7kQVQUIF8C5BcACANMNZwUB8hV
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+
+
+
+Valin, et al.                Standards Track                  [Page 273]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 274]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###hcyc+rOgmoGvBfQ9F5kEs8kk3RHI6dXryYRFMMc6x+CNSunBUPZe0DFWJhjKMyfs
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+   ###90YSAnzjNh2lfjWfHyXjwL8VySC0OvaW+VtM2W0GT3HD4rU1vojl7lOUrWETZ/3D
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+
+
+
+Valin, et al.                Standards Track                  [Page 275]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###aJixmMGqvehtTM6RSjuzB9eKdeqn2S0i1nrFHFSuzNpa1tbQbYqJCpXKG/AWKnNH
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+
+
+
+Valin, et al.                Standards Track                  [Page 276]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+   ###cCd78k7tiDl2nTNTi5KntMZXFL6hi5IY3kga8BtDNRa8d84AHjn8+ADs986VfDQv
+
+
+
+Valin, et al.                Standards Track                  [Page 277]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###mAiZyfQuTH4fh7RfSt4LJlP0ADturK0BPIDSF2KDaRtIq7PuKiMPtMxP+ITf9MeC
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+   ###/4c1VTJ90FW79q+2UTdNvVqKg5dpDgyM9K++qlYb8KPfBAkDrwuaFOpfh4fYE82Z
+   ###j3qW42IZNGSz43AH6C58t9/aC89PTt5rzdIpHB5zA1vb6qziN7WpZrUrUVD5WWuV
+   ###5dMa3o3QwWscX/VjFhGDCYWAIfbIj4M4Z09NhKQ5sWrS6Wwn1peYeGEwvi4XyFm1
+   ###IDOxOXR4Vp45tMhNtSCPzhYpfFYCJ+a5OYbD34H3Eic8Shey+uS0V66UV670VmWe
+   ###+ugnT0nVW/L2jCQ03mpxV9s7zHv+lWVErlXLxtBf43HCHX8fMBWMQALVzHtu9iJX
+   ###/idnUq/ESPxUQlbyCqZtJ6dZLTa0K69TXg6nZIKlC7oohD08c9Z7OT7MJ8/Iy5SX
+
+
+
+Valin, et al.                Standards Track                  [Page 278]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###i+kWxLayhw5cf4hUTIn8EJLGAF8xCxEor+iuhOubkR5pF4hkML0re8Qgxh0gG5/8
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+   ###+upWsoGyVnBtWx9aGF/lJmI1UWgP8o9VUWqW3426XShPd1UXygCvPGnhyclkdnzH
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+
+
+
+Valin, et al.                Standards Track                  [Page 279]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###JSY+7W7h3zf4FD8ZZggGDjR70Rt8xoYVJZ5em5ea0B/zRLI3P3vMTWxHOIlphioh
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+   ###+4r0OwVJdYOb4i5OQ9isILt244ewt7t38LmGaoW06vnrm0wo3uR/verKt6287sg7
+   ###BXsKq583tL68dZ/10Wiyv/Ua+4uenxR6xJ9LL8bm3GP6n/2mPgEOuN9ssL915k3l
+   ###c68qvyomyAfb4N5W60qJt6YAnP+tcwBr/G/AJ+I3OMAbcwMefK4FGuBBnQ0WBKzT
+   ###wGff5YQ21tnfdT6xJp9QoyYB9zmgPgfU54D6vph8w5g8B3yjmk5+nU+S9+811vlf
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+   ###8+sYThOAED4b9zvAr8Kzhk2rfCn9Gsc6n53822jof9dr4m+RJRfDWuTqNQU5ipXi
+
+
+
+Valin, et al.                Standards Track                  [Page 280]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###/fsbbnjq6Xe56nxl0r8N9/cm/95siu/zwG5RrIQtELDWdRitvw35N6XYDFiz/so5
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+
+
+
+Valin, et al.                Standards Track                  [Page 281]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###xmQj1vi/DMbXxrtWG2AGF0NihsQTiB/kv8GtjtB32XsIJybAjB+ws9Vx3qtH/Lb3
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+
+
+
+Valin, et al.                Standards Track                  [Page 282]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###7s3z67D55hi6FDOS3YR6K7vigo51zKpojUajfk8wXgLwCQE7Lx/KWUE6rFQpX6R5
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+
+
+
+Valin, et al.                Standards Track                  [Page 283]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###aZmEPA4n/LTzneP/g8Z6rWnmf/TXa8v7v+X93/L+b3n/t7z/W97/Le//lvd/P9T/
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+
+
+
+Valin, et al.                Standards Track                  [Page 284]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###5JTqkGQE62YE4W65Z3T1V5hJeZ6ZFOn2lMr5tHt/xOFd8jwEdYa44A+9yWMWooyu
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+
+
+
+Valin, et al.                Standards Track                  [Page 285]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###R/u/bM3q5uIGJP6bYb8rZORtR5T5RllR3nM6k4e4ImrbLyWgDAAGrcbpSxv2S3B0
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+
+
+
+Valin, et al.                Standards Track                  [Page 286]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###UNXKUdUxkKmlmhx0fQ4OanAUebHQtIuMWxzU0DDFjhaUrHHQhrGb/Iy+NvU5KWd7
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+   ###5agF7dOeUCwIz5qW4xbOG+16W/arTDVBaLbciHkp21AHzOp3j3WBSik2NN+oDkoa
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+
+
+
+Valin, et al.                Standards Track                  [Page 287]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###JmipQOhmOU0RdEChrpZZxyTFzAuv+nAA/8opJWYBdTgYxHQi40mcOyKb/Au5Qz5n
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+   ###tH14Qy1XLF2XYnKhG9TzZ4HCWT9ONmX93scIFEhMh92ZTiqYWRhor4LpjIW7rhSO
+
+
+
+Valin, et al.                Standards Track                  [Page 288]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###5mbfsJ6YjNbNwmHoo+G95xxpZsdpatyQmEs4GG4pHe9yXMHBByQDbBBrhzClNJLz
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+
+
+
+Valin, et al.                Standards Track                  [Page 289]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###M9vJ29vZ4XrqjaZgRqdHu6nntM6Ido9/fhLTgffC+KFDUYEyyjEv9NKFZOyEPJqZ
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+   ###//e61R4NDoKryc28SsCn3//V6n6tHvn/ra/h/Z+37i3u/xb3f4v7v8X93+L+b3H/
+   ###t7j/W9z/zV7//fXu/sVZ8+Bw7+1Lrfy7+XSJbEri1R2KQOx60m9zPkZcBwu090PM
+   ###N8VtkzxkjFEHVBfzGUqBu6H2XbvV/9uYmEaf99wRvJLONv72fjD6xAcQPodHxz81
+
+
+
+Valin, et al.                Standards Track                  [Page 290]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###SQajnFfPbpmYCreZin5YgUNexGc/qIfc140mtn/29vSyCF/S2Fyw3T048OpLGNYF
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+   ###8jflxIi1i/wIvviTIvgxFFz+phSs39rl0pvEBd/MxiEHSnUzA8CLdy93rVF7wKOc
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+   ###r0rwqeigDA/xvcedWWawDxiXLvDAw+2h9160qSPTgMCXIKfWKBsn6I4JKybDQ/+A
+   ###FdMNYvNasfiVIB/5KsTUMpTLAyC+7g1amFSvFw5KLITJDIJQxuY/e/aM5/TWlwO+
+   ###z9gz0MLYKg49t0W3MFHAiQExur6hHz4gnNH8ccJl+McyzK2yPcwkMAquJ71SNLmR
+   ###SBB+Pxr0b9j9bdAXCQSDXyeYUGU8UBc3P/2EFzdLX+07BeiGE0QGtal2BqFlz7fy
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+   ###8z5omZd7/GI+BfJMC6LWMh34+K2Lfjv9jcDPXH+tZTr4DqdXPQDw0eBb96rqzvCU
+
+
+
+Valin, et al.                Standards Track                  [Page 291]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###nwMYcFH1m/E7RldDr97Mvrm86/Yzm7S+ZDXBxJBLxHV4+QX+D6U81VOz4A5d1On9
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+   ###GzgRRHr1Kvv4gZ02TwDPJ43mm5e7sDirbIN93MxTS8yvZQ9GdVD0ITHn9sHhyeVu
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+
+
+
+Valin, et al.                Standards Track                  [Page 292]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###q6NlcrsfXBulMIsxqQ+gbA8fCiZ1G8NKhoMPCWw+9nIy+NMAE9+0RLFhjt2aTn72
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+
+
+
+Valin, et al.                Standards Track                  [Page 293]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###YuMsU7dYrHY4HiEGRDKgEr0JeU0s6xvxnE8cjorsyQP4yR848qQtJaVFwZxbYsaA
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+
+
+
+Valin, et al.                Standards Track                  [Page 294]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###5hRj89ImOCj3Fcz7w7/+bUYm+U8ro9XsXxO6vM2Zvp4iOaDja+UfRopAIZ6rqMT+
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+
+
+
+Valin, et al.                Standards Track                  [Page 295]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###g2KpJRKg2K+TVn/c/Tfna10YoM3Zi1myatr+I/OLIadx40vQxqcJivMwvMD6zQeY
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+
+
+
+Valin, et al.                Standards Track                  [Page 296]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 297]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###oKQhYVnrPOVAAwUCr0R3diJ/8Ni6pQBQyVbxda8Kbf7oc983fTJHwR0GSaPIf4OR
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+
+
+
+Valin, et al.                Standards Track                  [Page 298]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###TIBu729BzO6j4OXiXCSe9FFC7qEjOXEaEkxA4O7BW7zg6kNby8tdbL7M2njLreYb
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+
+
+
+Valin, et al.                Standards Track                  [Page 299]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###6Ynhr9VwbIdVXEFdOrjA39ErTAFouxbRTudlDU6A/Z00dw/+hw/p8JTTx97KHnv3
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+   ###mMzvJvuqYC63VJSTwYAj3fdMDIpZ+5UN8W9d/MtH854/F/+KvzfEv/UIcq8msCsz
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+   ###5iDyQ3p+TVBFRCkRZYh/18Uqyhx8MtvkmpjDmqQYbZ5V8U1VrG7Vs3Aj5u2L5zL/
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+
+
+
+Valin, et al.                Standards Track                  [Page 300]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###GCVrqFeNfhc+4v9x+R/fvZmX83e2/r8Oby3/b79aWV/o/wv9f6H/L/T/hf6/0P8X
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+
+
+
+Valin, et al.                Standards Track                  [Page 301]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###VyR9qvWvW/z+eUmvE2GUrhHWSll5Q+FU7k8JX61qWh3XNfud2qc1C9Z1i0c/t2x0
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+   ###Y7MJwHK3X0rSoI/7xsjXvQHI7TD+ql/MBCFHWexu/wTm4h76VCnv3LrD1zLMO2vb
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+   ###j9nNPVCCKkp/RiTQYQpo/9yCMxmj4+l4D4kO8WSje5QK4xKXUUha06twwnGVCrrf
+
+
+
+Valin, et al.                Standards Track                  [Page 302]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###H/Q/AzR4smIvutYJU7a1qkKEkmK3/4EKK5MC5VUMnR8NDb1eeQhTxWrKJGKQPelz
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+   ###24xECP7Fllsw0THJ86TIGZb5h0nqhdE1cAjzokbvXWwq2a/RrbNrIz3KpM9VyI5c
+   ###DysTjClgWTkjJNsY66LB3CWD49N3hxeNw5mkA28q2cBxiH/DgzzzMI/O8oTDPP8R
+   ###rvZsMBo13+A0H3WEP/4E//Ofjv8JZ+Ms52KEMk4aMNBmfhWUQ1P3HIe0BdCcj0Uy
+   ###dAs9HkcQhBwLNfS2tvznRenkJoSAK/s4oTNL1+DeuNyPY3PKdYq9V7xYbbZpTrF8
+   ###uM11iJldJZ1c23/eowt7/hYHl8s2+jXy0ad/ycIXs6viMIsr/Ufe/79utUeDfbQ5
+   ###z8sFIP3+f61SrVet+/+q79cX9/+L+//F/f/i/n9x/7+4/1/c/y/u/2eK/xR38y9x
+   ###jVBU3D0/Bt1uH/46e3t6aVzSp7SRYaFb4bjTHTy73YmEVKGBU/smfaC6/NwaHfDf
+
+
+
+Valin, et al.                Standards Track                  [Page 303]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###tHv3wTBs8gt9SrstL/1dLeCt2/YCjbhC0/ockKBWEGI8F6sLqoci3RNld3IBR+/Y
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+
+
+
+Valin, et al.                Standards Track                  [Page 304]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###qDE/UNnUSI3mQY07OTcuDJhNjtRoHuSYBJUh71COnUyKUq3cNNXDTGOe/cCP5WtN
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+
+
+
+Valin, et al.                Standards Track                  [Page 305]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###4fH49ODw/PAUL3wtc2jGOjQRiCCRBq0WFiVOgW6tlwjf9Cz98DBA/0GbOciXyXBu
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+   ###tkY3wRgdt5tXwzAlXS21Q8MLpkrE+zBWgA+Ked1/P5VEkVpyjjk9awLNn13EXDiv
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+   ###glRxetHs7MVTMegJtfT0FLbiqnAsMaWNRtEDzdO9hAxtdueeP0Pvr/cykptld/F+
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+   ###b/eBeI4viYi8SvN1A0kpCdajLgmqHFLa9LCmBCJNnrsmyaRigTNrhaj6pNMunNvN
+
+
+
+Valin, et al.                Standards Track                  [Page 306]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###xj/ceSscxLy1HeGbZ6VwWZvEtjXyGxXMm2IbH2a3othPnRVLS2zaHwtAV9cu1xK5
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+
+
+
+Valin, et al.                Standards Track                  [Page 307]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
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+
+
+
+Valin, et al.                Standards Track                  [Page 308]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###JVXbslhoWbJ1T1FG1IEAp8whwJytR20CAIfn4xKdqNq9siCP6FvWj9/VgYuqz7/n
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+   ###raU6s2d1Jnd3D5tmo6QYYmqrUZS8Z/Fs3rw7GQ/u6Aq0MDJ0xmIULaCpCS4eHN9A
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+   ###X/N+1Z5Cf/ZtTjYVJXAAXMHOL+h0TrUiIiFttQxb/q+mWqfFA5scIAk49IIs1NZW
+
+
+
+Valin, et al.                Standards Track                  [Page 309]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###ksKKi6tr5sbEyN09qneP6ZqYCgvAc+j/89inq9XQGtptvnSDVt4ml1O7Dr3LgqEd
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+   ###LVXTymbwQ8yomdSpnZlUynWU3X0o6WAS8vzXwKuA98r0ZWP0G2pdB3hmrwJ0jCdq
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+
+
+
+Valin, et al.                Standards Track                  [Page 310]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###TYNQ3Q7DDdNP+6JAYoUVTWMKDhGX5K65mR73PqwKrMcTjUczyoD2Qlrusf7XpiW3
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+   ###sTJw3WvtkNsiaojXJlNfFrDWdIIXoBXiuDS16yBW6XW4C3K/SuYoWlrg1T1NcTY5
+   ###JUR8zc0lF+ZkYZiepoRlFEj6TReU6CZlLXMsEXXxqNUpeM8q16tVf72+UVzWIndj
+   ###qyP2wLdZI3t5uFjkLP0qctSX2LNnz/Tq5Mp2lFJK83Or2cMsdK2hOKr5LEVZW9WG
+   ###tPUC1lAXg0mZ5pHF+YSBriB7tR1blPcQCvpmNZc3bwGTLxIM2loczudWb4KCCswC
+
+
+
+Valin, et al.                Standards Track                  [Page 311]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###hAmag+Z7HzdzYkp2/k1a+ebkFkbhZiPffvI3abUH5Q9W3YtnrrTKQtMQxQxHCm2J
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+   ###0qlBYQPydcnIoYwxnRpib4Jxsxdcj+UQaZjl0RHCstHt8D/RU1ZHOUcS2pDwzVal
+
+
+
+Valin, et al.                Standards Track                  [Page 312]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###yP7OsBw6/rXiRah5FF62txUUy3584t3YxMt85tbESRb7D575ijf73LEs7H/s1FM2
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+   ###HOoN+7vrSGIvHNxf40om3ra2Z3CPSMKgpukY89ft+xo/Mzc6/CVv8HC7xl09oo0I
+   ###/O2H7YTFLrGnxtgx7wDjre7TUE54J1ngjgbNjDhzOr9pWKPjZKci65H9sP34IdOX
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+   ###AL1y0C4vnblg29xTsGkXK+D1Bv0bs3w7d8wpEMfs53Gzo+i0Iq6VEWpk+sTldBLD
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+
+
+
+Valin, et al.                Standards Track                  [Page 313]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###1A6c6bgwyo5+zXYz0wuvfs0fA8DPwG8jFNsuSdMJxWJeGdI8airaGU8VCd1nfFHi
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+   ###blufA0TozQC2cf9B5KJgD6gugzyGTbAASGeA92yCMv4djAZmxLh2IisIlw31wxnY
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+   ###DVoYoKKnosQlefA0TJg8Q6OzHdqo8Izg+cE+zYp2xk10widFL9QxAkNbCQDieUz4
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+   ###tYmOs0j5Sv4utpfNJkUmV9EgqAgTKsuRLK/m7saSmFMpy8AzMVM8XdXOLCqiUzw0
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+   ###Wvxn/4kGgBnNiq2Km4pVZdeJ0/RXi7kYp4jJib4umaSIpz4iZ/enAokd2qxddCmD
+
+
+
+Valin, et al.                Standards Track                  [Page 314]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###6lUflqSVh56BoZEFcDgatIMwxIM4YlNuAosxod4gHDeve60bGQXckXt1052xocnD
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+   ###1rtQ+vJeoOYEbf2MzQIGadvUm+CwNpsZmbLGvIEWtaPg+z1DYWMabpQLcDNlZi5p
+
+
+
+Valin, et al.                Standards Track                  [Page 315]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###F/bGaTg6ZijNUnySkIqt1Qq3ttjBHm5t7hqDVF6KvIjDhCaacSU0g8k050UeRFoC
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+
+
+
+Valin, et al.                Standards Track                  [Page 316]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###3KKOC1BshNUWIVcGdAaZnxhNRdTATV9jCG4fGqxZNnfWQjyoBagMo9XHbIbntVZR
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+   ###/28+bW5tof//1kYz3H66Bf7/W2Hw4P//N/L//z0eX6+9mVwx/gTCrqMb9YNH/4NH
+   ###/4NH/4NH/4NH/4NH/1/q0a/CkXEPvHcvf312etT+0fLNV6+1I/8qO5iu165XyRtg
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+
+
+
+Valin, et al.                Standards Track                  [Page 317]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###o/F4JMgwO7IYBVUnB7dtoU6likX/i51WkUiDDCfCJWTaYQ+9mwloY0DTC7enO8k4
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+   ###Ybzu/Rfwha6+bGiwU5aAoyrh8VXA8UMGOH4oCw64pgIgcexBdoCx+rhxyX4KcMfm
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+   ###VxnSsYXFcbF9Jwzt/MKRLwLRay/JeNgHkGNMF6iEIIpFPAkVFySumc00Av9Jw3xl
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+   ###wwSPw3lahliYTMCjbfMTmoZ6h4NuOoK0jJMKzecc22qY3vXZ5rk8/thDExKhxwcH
+   ###4eaefAWWC6/C3gXbPkbKV19C+WXH+rCBrYh3Ogo9HQn0DkMED/qcIUm/cDoLH8e5
+   ###vx9ugkYy9Qli/7Ohai2lo0zIyng7UN2zP22cpwLnc19/mVSRH1kV4vEPJ4X0beDK
+   ###LVZGmh4vvQopZuaod5+GcJ7HCT9/MAxKxDcWnsnQskrs6cw3gpftHeMXJyQdv7gN
+   ###UDR+UmyR8UvCMNf4+QWK/oedNqOr07NzhhMMp2TPn8yrXfXslLZ1MO1XN+pecF9f
+   ###sHK4ROU63DlZqPLmzhKVxZzDZeYcLjPncJk5h/d4j0+iQBAuhAM60fMCOGBXDpeo
+
+
+
+Valin, et al.                Standards Track                  [Page 318]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###PN8y0spz44BjzuEycw6XmXO4zJxtHNheCAdkRuwFMMCsGi5cdb4F1FXnXvvUXMPF
+   ###5xouPtdw8bmSNb++607ii3BzoVU3kqGLgZWHo1Ubx7Zwbew7XKrvFFQ2FzsTZX75
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+   ###cVV7iT0eLrXHw6X2eLjUHg+X2+PhUns8XGqPh0vt8XC5PR4utcfDpfZ4uNQeD5fb
+   ###4+FSezxcao+HS+1xnLhxxF//+4ud8uWHttw2Wg4XsxZUQkVlccIkTkJNw81XUleG
+   ###GqnJRGrhoEA8PI77Ud2rjm6m8It/4xlaIUEr+5t94W/Ba+tIWFGq0bCnMqHyTyoq
+   ###xUVEP4khJfoJvOHCc/1MQ1CY4UWJ85ewQbS740TGoXX4i6l27ShBqH1PNSrNCG2d
+   ###MAcGdJNEbfTIlS+Ipr9NWkJkVPp4s75Sq5uvL6YzMnHQjZszxiYND2B4I66PgHIa
+   ###7gb1UhNBR8VWU9xgaHd60xapTnIfQjxj+imZjtSzuNuAq64WHZdPGzV5ksZnb9/+
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+   ###meIZHo5u+pA+GcKxsungTVke4seyAEPf5e15uCrKqcMyxYvURKqxyySKPlTQHaxZ
+   ###906Ojv67ffT6he5M8M2X06jfh2L6S0pbhgDjV6ITCGWKi4bzgJd+XpcndG0cjiLQ
+   ###wLqICo6uTP66FN/X7V2ZPcATbBMMrJgmDHMZwUBFaGYcqt25b2CF8KQz0CKQaAGx
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+   ###e/jVqT9D3XuiU0TLS7GTCRjIRVRiRW1yXSUOWdsMHrxnGfJBOUvg6CAgA9BUaN83
+   ###KKETLPo1rMncbhQmQKZ90CvUdf6w5y9P3z07PaoQsknGVFRZ2rMqyhJVvvKvz99V
+   ###hFl3rkoQDIadzc9evj6t9OaqDKf3TwCtPypag1m++ktM594+PjqsGArG8i3wZG0Q
+   ###OOn10U8nFUPBV76VF6e/V4S+qnwlfrS0IQxR++3Ru8OKrTtSHmX5bf2Ajbz572c/
+   ###Hj17UXkC3py+n/Lmsq+T6z1zEaldKvVdRbv0a2xM0fdftjEh2KOnokqT7UOghaGa
+   ###M63+uytpO7PwDmQAXtURTFbJyEiMXVfr2shf0PoguohvBnO2Lg3sBW2z52jOlg37
+
+
+
+Valin, et al.                Standards Track                  [Page 319]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   ###fUHzyQ3D9gX6kBb7guZlrJjClsENqKAtSD2a0Q6N7JzdgPDby2iDHKupU9W1g16I
+   ###IA88Wt9jdvL/+Kc88yuPL9SO9fM43Up/NLzyM/a62EQkDnt6FEciVIMcAOdHO1Pv
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+   ###eFqnjAJPnqAGuuI/DpuGa4HZvegREgHw0tu2jh7PG1HMdFCwlBVAqZq7yqs/7bKS
+   ###itiu0yFk1ArzaoVZtYK8WhtZtbZyKm1mDnAru9KWruTw3HBVMx0ZBHbJFdpPuZnA
+   ###GhNy0rjsdJOKsoP53j64jASh1bdzKTUmVFth2n8pawwB9yAmOHzgcO7Jw5rSTDAm
+   ###MjYvnPkOZxHjIolldbnP9JAp3IP3K+lfdNsxIqM33EbOhisvMojGNqEtfy+jf9c5
+   ###bPs2pCNRWW5DE7C8X4JDMKNxdS+oe5t1z2ZdeX+XQAjhk8N5MY3PcG2O32ECekvD
+   ###Y/WuHY2T4gdpNdnnz7S9/UKsSjECL4e3nT7EzBRh7lhrV9PrXaFpI2070Kpws5YF
+   ###on1tqyR4aPN4gol62BEZeta6sfr7dIbzwu5dB7U1EA8Z+ad6GS+xaDaOelNkvYTa
+   ###zLtiElWhZjOFPHVDoCu3KM7NjkmT0tBRasVHPAndcvRKKzlUu2fqAtk54z1cdKs8
+   ###DXQ22ly8UaVdcDa84WzYIJjOeuF52d1yKcJMGhxhVUvz9K5tGrnVXQZIsSEvChOH
+   ###KJkWQxZTDaeTmWQstLh3mwZC0uJDPwurcQ0i9O181i/OXeUrlQQDCB4f+/+CbCvs
+   ###aGzwf9OlY8i/0UryoAioq+e/T8WDov3NE4a4oLLv4NbTsKKDTN+xwNkynjjt+HCf
+   ###cz4QbOWICqp0uvom4UudEC6/a8kMPWoVnRf0hhbkX5E1UzNzW5SyYS0SqpHmD7zs
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+   ###oB0jL9BPHr98/eyn9rtnr384qjzJOKH9fIYFzIMlT1WPBpfUKC8zRkYqW14+S5Kh
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+   ###MX0ERKY3r0wutv50Mhm743/pxPTYCphqOVClM/Lp5vzmg89kbrl2VWbltaQxHm6C
+   ###DJyJliBWpUw7B00izYKMv5/yHE6tckXkkTto5SjfyDGCgjrkBhoNVSjrj9fRkNUG
+   ###iSEeNmQIb8dRg/2Z3Tj7kUV1fI2sUmJUnek0Gowh8oaw+HC1Lx0Nj/cezWSME9fw
+   ###1LljLIdpxwLZnUf+8P6JaXG9XYvDMVBwBEKXqU1leOrvOfvO1HSQqU6iq5t+ZyLn
+   ###udQsuFK1YYycPDrH3swa+8pcs/naIHYN837FxeKZIzlw8D5ZCJqqi1bS7OVziLFO
+   ###qcdqtoHNFnP12UMVzCNG3Un28oq0WNdnca0CXaoe/fO8OkR4aiUk+HpRaSZ7tSqJ
+   ###FbG9FGKlj4EiYdQBUOfp4IS8nwfURU6MeQ6PQkDcl0TOfY6c8LfXaOWeOIaQghWc
+   ###omLe/k6BBQ3uKn48Z0SlVb5QpyRFAWW4N0fvF+71TN7xkZN5lP0pzwUmeiBNcose
+   ###VrwmOhxGsMEFBa5bYgkPS6gUMRMGittO/4anLJNv4ciSSV6s+AxuPu5fxpl6zqRm
+   ###DAFCJ6fnBh+QtsofeMIa360JwR4p1XHqmE6LTsLL9B22y6cI+QEibxAngw5k3i7Q
+   ###UK5Kf0sXtIrqxkNxc/Zx/2LXa84e7/Rn3m0ifuXjIjcLI7eUV0qJGli8FNjKN2dj
+   ###mqU3yxIts8XKAo/cFcqEykhn8JtkQUYUv+HZgox4IvhcMzjXKrl3IEO2gUnXLMNE
+   ###T/XRZyyA+XFXfdxbWYCPHU7AZaK51kzLO4+cZpkcVZz3ASkj+4dq8byqDi7ofajV
+   ###MplXGAqDTzw8Yy2cs2oVHtfOF2/KUzVlQK9go+vsX+d42FB8Bt0g2nIOSsbNy1k1
+   ###fTVYhP+zRXlzUAw5foXkTsNeilXFsHx2T/T39qbLfIYd005VtL2K+FUzMcD3nkh1
+   ###5v2Kmyh1bqNJ50qln9o1Zvr4qXQ4yaYPQdTYqMKELKNthWhl5eLiBHwze5DpaI5Z
+   ###nXLG0v0wLjUY2CPWgAxXlFy8d0CpN41vM4BUACU9JNZIWRhB2Ww4iWHA6TG86Ezg
+   ###DLiNRciyvNHgUJJ/T6YVRMB1TjYaiI44xPUKz2turVcRICEjL1BZdh7doWt0AtLo
+   ###x+u4B3LoNJqALx0MeBIlELjK+xhNYPSjCSPuYmtgGPoXR89/+aF9cvrm3VH78Kc3
+   ###J0dtCJJ6wt0I4/6H9imkYD1hOFsR6QjjAbCa05m6SGLoTS8g8tzoDlRA9LPkbeRn
+   ###9ixhPYmEYkUl3AMsMTeVdvFWpZXnd+ocSp8/WEt/hyfr4yU1DtBj6uSXw8OjkxNI
+   ###jfR/Hv48/Hn48/Dn4c/Dn4c/D3/+v/nzP0cUzKcA0CAA
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 320]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+A.4.  Test Vectors
+
+   Because of size constraints, the Opus test vectors are not
+   distributed in this document.  They are available in the proceedings
+   of the 83rd IETF meeting (Paris) [VECTORS-PROC] and from the Opus
+   codec website at [VECTORS-WEBSITE].  These test vectors were created
+   specifically to exercise all aspects of the decoder.  Therefore, the
+   audio quality of the decoded output is significantly lower than what
+   Opus can achieve in normal operation.
+
+   The SHA1 hash of the files in the test vector package are
+
+   e49b2862ceec7324790ed8019eb9744596d5be01  testvector01.bit
+   b809795ae1bcd606049d76de4ad24236257135e0  testvector02.bit
+   e0c4ecaeab44d35a2f5b6575cd996848e5ee2acc  testvector03.bit
+   a0f870cbe14ebb71fa9066ef3ee96e59c9a75187  testvector04.bit
+   9b3d92b48b965dfe9edf7b8a85edd4309f8cf7c8  testvector05.bit
+   28e66769ab17e17f72875283c14b19690cbc4e57  testvector06.bit
+   bacf467be3215fc7ec288f29e2477de1192947a6  testvector07.bit
+   ddbe08b688bbf934071f3893cd0030ce48dba12f  testvector08.bit
+   3932d9d61944dab1201645b8eeaad595d5705ecb  testvector09.bit
+   521eb2a1e0cc9c31b8b740673307c2d3b10c1900  testvector10.bit
+   6bc8f3146fcb96450c901b16c3d464ccdf4d5d96  testvector11.bit
+   338c3f1b4b97226bc60bc41038becbc6de06b28f  testvector12.bit
+   a20a2122d42de644f94445e20185358559623a1f  testvector01.dec
+   48ac1ff1995250a756e1e17bd32acefa8cd2b820  testvector02.dec
+   d15567e919db2d0e818727092c0af8dd9df23c95  testvector03.dec
+   1249dd28f5bd1e39a66fd6d99449dca7a8316342  testvector04.dec
+   93eee37e5d26a456d2c24483060132ff7eae2143  testvector05.dec
+   a294fc17e3157768c46c5ec0f2116de0d2c37ee2  testvector06.dec
+   2bf550e2f072e0941438db3f338fe99444385848  testvector07.dec
+   2695c1f2d1f9748ea0bf07249c70fd7b87f61680  testvector08.dec
+   12862add5d53a9d2a7079340a542a2f039b992bb  testvector09.dec
+   a081252bb2b1a902fdc500530891f47e2a373d84  testvector10.dec
+   dfd0f844f2a42df506934fac2100a3c03beec711  testvector11.dec
+   8c16b2a1fb60e3550ba165068f9d7341357fdb63  testvector12.dec
+
+Appendix B.  Self-Delimiting Framing
+
+   To use the internal framing described in Section 3, the decoder must
+   know the total length of the Opus packet, in bytes.  This section
+   describes a simple variation of that framing that can be used when
+   the total length of the packet is not known.  Nothing in the encoding
+   of the packet itself allows a decoder to distinguish between the
+   regular, undelimited framing and the self-delimiting framing
+   described in this appendix.  Which one is used and where must be
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 321]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+   established by context at the transport layer.  It is RECOMMENDED
+   that a transport layer choose exactly one framing scheme, rather than
+   allowing an encoder to signal which one it wants to use.
+
+   For example, although a regular Opus stream does not support more
+   than two channels, a multi-channel Opus stream may be formed from
+   several one- and two-channel streams.  To pack an Opus packet from
+   each of these streams together in a single packet at the transport
+   layer, one could use the self-delimiting framing for all but the last
+   stream, and then the regular, undelimited framing for the last one.
+   Reverting to the undelimited framing for the last stream saves
+   overhead (because the total size of the transport-layer packet will
+   still be known), and ensures that a "multi-channel" stream that only
+   has a single Opus stream uses the same framing as a regular Opus
+   stream does.  This avoids the need for signaling to distinguish these
+   two cases.
+
+   The self-delimiting framing is identical to the regular, undelimited
+   framing from Section 3, except that each Opus packet contains one
+   extra length field, encoded using the same one- or two-byte scheme
+   from Section 3.2.1.  This extra length immediately precedes the
+   compressed data of the first Opus frame in the packet, and is
+   interpreted in the various modes as follows:
+
+   o  Code 0 packets: It is the length of the single Opus frame (see
+      Figure 25).
+
+   o  Code 1 packets: It is the length used for both of the Opus frames
+      (see Figure 26).
+
+   o  Code 2 packets: It is the length of the second Opus frame (see
+      Figure 27).
+
+   o  CBR Code 3 packets: It is the length used for all of the Opus
+      frames (see Figure 28).
+
+   o  VBR Code 3 packets: It is the length of the last Opus frame (see
+      Figure 29).
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 322]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|0|0| N1 (1-2 bytes):                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
+     |               Compressed frame 1 (N1 bytes)...                :
+     :                                                               |
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                 Figure 25: A Self-Delimited Code 0 Packet
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|0|1| N1 (1-2 bytes):                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               :
+     |               Compressed frame 1 (N1 bytes)...                |
+     :                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                               |                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               :
+     |               Compressed frame 2 (N1 bytes)...                |
+     :                                               +-+-+-+-+-+-+-+-+
+     |                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                 Figure 26: A Self-Delimited Code 1 Packet
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|0| N1 (1-2 bytes): N2 (1-2 bytes :               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               :
+     |               Compressed frame 1 (N1 bytes)...                |
+     :                               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                               |                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
+     |               Compressed frame 2 (N2 bytes)...                :
+     :                                                               |
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+                 Figure 27: A Self-Delimited Code 2 Packet
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 323]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|1|0|p|     M     | Pad len (Opt) : N1 (1-2 bytes):
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 1 (N1 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 2 (N1 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :                              ...                              :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame M (N1 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :                  Opus Padding (Optional)...                   |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+               Figure 28: A Self-Delimited CBR Code 3 Packet
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 324]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+      0                   1                   2                   3
+      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     | config  |s|1|1|1|p|     M     | Padding length (Optional)     :
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     : N1 (1-2 bytes):     ...       :     N[M-1]    |     N[M]      :
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 1 (N1 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :               Compressed frame 2 (N2 bytes)...                :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :                              ...                              :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     |                                                               |
+     :              Compressed frame M (N[M] bytes)...               :
+     |                                                               |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     :                  Opus Padding (Optional)...                   |
+     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+               Figure 29: A Self-Delimited VBR Code 3 Packet
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 325]
+
+RFC 6716                 Interactive Audio Codec          September 2012
+
+
+Authors' Addresses
+
+   Jean-Marc Valin
+   Mozilla Corporation
+   650 Castro Street
+   Mountain View, CA  94041
+   USA
+
+   Phone: +1 650 903-0800
+   EMail: jmvalin@jmvalin.ca
+
+
+   Koen Vos
+   Skype Technologies S.A.
+   Soder Malarstrand 43
+   Stockholm,   11825
+   SE
+
+   Phone: +46 73 085 7619
+   EMail: koenvos74@gmail.com
+
+
+   Timothy B. Terriberry
+   Mozilla Corporation
+   650 Castro Street
+   Mountain View, CA  94041
+   USA
+
+   Phone: +1 650 903-0800
+   EMail: tterriberry@mozilla.com
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Valin, et al.                Standards Track                  [Page 326]
+
diff --git a/SPECS/opus.spec b/SPECS/opus.spec
new file mode 100644
index 0000000..c7d9061
--- /dev/null
+++ b/SPECS/opus.spec
@@ -0,0 +1,113 @@
+Name:          opus
+Version:       1.0.2
+Release:       4%{?dist}
+Summary:       An audio codec for use in low-delay speech and audio communication
+
+Group:         System Environment/Libraries
+License:       BSD
+URL:           http://www.opus-codec.org/
+Source0:       http://downloads.xiph.org/releases/%{name}/%{name}-%{version}.tar.gz
+# This is the final IETF Working Group RFC
+Source1:       http://tools.ietf.org/rfc/rfc6716.txt 
+
+Patch0: 0001-Fix-several-memory-errors-in-the-SILK-resampler.patch
+Patch1: 0001-Fixes-an-assertion-failure-in-SILK.patch
+
+%description
+The Opus codec is designed for interactive speech and audio transmission over 
+the Internet. It is designed by the IETF Codec Working Group and incorporates 
+technology from Skype's SILK codec and Xiph.Org's CELT codec.
+
+%package devel
+Summary: Development package for opus
+Group: Development/Libraries
+Requires: libogg-devel
+Requires: opus = %{version}-%{release}
+
+%description devel
+Files for development with opus.
+
+%prep
+%setup -q
+%patch0 -p1
+%patch1 -p1
+cp %{SOURCE1} .
+
+%build
+%configure --enable-custom-modes
+
+make %{?_smp_mflags}
+
+%install
+rm -rf %{buildroot}
+make install DESTDIR=%{buildroot}
+
+# Remove libtool archives and static libs
+find %{buildroot} -name '*.la' -exec rm -f {} ';'
+find %{buildroot} -name '*.a' -exec rm -f {} ';'
+
+%check
+make check
+
+%clean
+rm -rf %{buildroot}
+
+%post -p /sbin/ldconfig
+
+%postun -p /sbin/ldconfig
+
+%files
+%defattr(-,root,root,-)
+%doc COPYING README rfc6716.txt
+%{_libdir}/libopus.so.*
+
+%files devel
+%defattr(-,root,root,-)
+%{_includedir}/opus
+%{_libdir}/libopus.so
+%{_libdir}/pkgconfig/opus.pc
+%{_datadir}/aclocal/opus.m4
+
+%changelog
+* Tue Nov  5 2013 Matthias Clasen <mclasen@redhat.com> - 1.0.2-4
+- Apply two crash fixes from upstream
+- Resolves: #1017240
+
+* Thu Feb 14 2013 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.0.2-3
+- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild
+
+* Thu Jan 10 2013 Peter Robinson <pbrobinson@fedoraproject.org> 1.0.2-2
+- Enable extra custom modes API
+
+* Thu Dec  6 2012 Peter Robinson <pbrobinson@fedoraproject.org> 1.0.2-1
+- Official 1.0.2 release
+
+* Wed Sep 12 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 1.0.1-1
+- Official 1.0.1 release now rfc6716 is stable
+
+* Tue Sep  4 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 1.0.1rc3-0.1
+- Update to 1.0.1rc3
+
+* Thu Aug  9 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 1.0.0rc1-0.1
+- Update to 1.0.0rc1
+
+* Fri Jul 20 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.9.14-2
+- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild
+
+* Sun May 27 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 0.9.14-1
+- Update to 0.9.14
+
+* Sat May 12 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 0.9.10-2
+- Add make check - fixes RHBZ # 821128
+
+* Fri Apr 27 2012 Peter Robinson <pbrobinson@fedoraproject.org> - 0.9.10-1
+- Update to 0.9.10
+
+* Fri Jan 13 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.9.8-2
+- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild
+
+* Tue Nov  8 2011 Peter Robinson <pbrobinson@fedoraproject.org> 0.9.8-1
+- Update to 0.9.8
+
+* Mon Oct 10 2011 Peter Robinson <pbrobinson@fedoraproject.org> 0.9.6-1
+- Initial packaging