Internet DRAFT - draft-ietf-avt-ilbc
draft-ietf-avt-ilbc
Internet Draft S. V. Andersen
Document: draft-ietf-avt-ilbc-codec-03.txt H. Astrom
Category: Experimental A. Duric
October 25th, 2003 F. Galschiodt
Expires: April 25th, 2003 R. Hagen
W. B. Kleijn
J. Linden
M. N. Murthi
J. Skoglund
J. Spittka
Global IP Sound
Internet Low Bit Rate Codec
Status of this Memo
This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC2026.
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Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
This document specifies a speech codec suitable for robust voice
communication over IP. The codec is developed by Global IP Sound
(GIPS). It is designed for narrow band speech and results in a payload
bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s for 20 ms
frames. The codec enables graceful speech quality degradation in the
case of lost frames, which occurs in connection with lost or delayed IP
packets.
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Table of Contents
Status of this Memo................................................1
Copyright Notice...................................................1
Abstract...........................................................1
Table of Contents..................................................2
1. INTRODUCTION....................................................5
2. OUTLINE OF THE CODEC............................................5
2.1 Encoder........................................................6
2.2 Decoder........................................................7
3. ENCODER PRINCIPLES..............................................8
3.1 Pre-processing.................................................9
3.2 LPC Analysis and Quantization..................................9
3.2.1 Computation of Autocorrelation Coefficients..................9
3.2.2 Computation of LPC Coefficients.............................11
3.2.3 Computation of LSF Coefficients from LPC Coefficients.......11
3.2.4 Quantization of LSF Coefficients............................11
3.2.5 Stability Check of LSF Coefficients.........................12
3.2.6 Interpolation of LSF Coefficients...........................12
3.2.7 LPC Analysis and Quantization for 20 ms frames..............13
3.3 Calculation of the Residual...................................14
3.4 Perceptual Weighting Filter...................................14
3.5 Start State Encoder...........................................15
3.5.1 Start State Estimation......................................15
3.5.2 All-Pass Filtering and Scale Quantization...................16
3.5.3 Scalar Quantization.........................................17
3.6 Encoding the remaining samples................................17
3.6.1 Codebook Memory.............................................19
3.6.2 Perceptual Weighting of Codebook Memory and Target..........20
3.6.3 Codebook Creation...........................................21
3.6.3.1 Creation of a Base Codebook...............................21
3.6.3.2 Codebook Expansion........................................22
3.6.3.3 Codebook Augmentation.....................................22
3.6.4 Codebook Search.............................................23
3.6.4.1 Codebook Search at Each Stage.............................24
3.6.4.2 Gain Quantization at Each Stage...........................24
3.6.4.3 Preparation of Target for Next Stage......................26
3.7 Gain Correction Encoding......................................26
3.8 Bitstream Definition..........................................27
4. DECODER PRINCIPLES.............................................30
4.1 LPC Filter Reconstruction.....................................30
4.2 Start State Reconstruction....................................31
4.3 Excitation Decoding Loop......................................31
4.4 Multistage Adaptive Codebook Decoding.........................32
4.4.1 Construction of the Decoded Excitation Signal...............32
4.5 Packet Loss Concealment.......................................33
4.5.1 Block Received Correctly and Previous Block also Received...33
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4.5.2 Block Not Received..........................................33
4.5.3 Block Received Correctly When Previous Block Not Received...34
4.6 Enhancement...................................................34
4.6.1 Estimating the pitch........................................36
4.6.2 Determination of the Pitch-Synchronous Sequences............36
4.6.3 Calculation of the smoothed excitation......................37
4.6.4 Enhancer criterion..........................................38
4.6.5 Enhancing the excitation....................................38
4.7 Synthesis Filtering...........................................39
4.8 Post Filtering................................................39
5. IANA CONSIDERATIONS............................................39
6. SECURITY CONSIDERATIONS........................................39
7. EVALUATION OF THE ILBC IMPLEMENTATIONS.........................39
8. REFERENCES.....................................................40
8.1 Normative.....................................................40
8.2 Informative...................................................40
9. ACKNOWLEDGEMENTS...............................................40
10. AUTHOR'S ADDRESSES............................................41
APPENDIX A REFERENCE IMPLEMENTATION...............................43
A.1 iLBC_test.c...................................................44
A.2 iLBC_encode.h.................................................49
A.3 iLBC_encode.c.................................................50
A.4 iLBC_decode.h.................................................59
A.5 iLBC_decode.c.................................................60
A.6 iLBC_define.h.................................................71
A.7 constants.h...................................................75
A.8 constants.c...................................................76
A.9 anaFilter.h...................................................90
A.10 anaFilter.c..................................................90
A.11 createCB.h...................................................92
A.12 createCB.c...................................................93
A.13 doCPLC.h.....................................................97
A.14 doCPLC.c.....................................................97
A.15 enhancer.h..................................................102
A.16 enhancer.c..................................................103
A.17 filter.h....................................................115
A.18 filter.c....................................................116
A.19 FrameClassify.h.............................................119
A.20 FrameClassify.c.............................................120
A.21 gainquant.h.................................................122
A.22 gainquant.c.................................................122
A.23 getCBvec.h..................................................124
A.24 getCBvec.c..................................................125
A.25 helpfun.h...................................................128
A.26 helpfun.c...................................................130
A.27 hpInput.h...................................................136
A.28 hpInput.c...................................................136
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A.29 hpOutput.h..................................................137
A.30 hpOutput.c..................................................138
A.31 iCBConstruct.h..............................................139
A.32 iCBConstruct.c..............................................140
A.33 iCBSearch.h.................................................142
A.34 iCBSearch.c.................................................142
A.35 LPCdecode.h.................................................151
A.36 LPCdecode.c.................................................152
A.37 LPCencode.h.................................................155
A.38 LPCencode.c.................................................155
A.39 lsf.h.......................................................160
A.40 lsf.c.......................................................160
A.41 packing.h...................................................165
A.42 packing.c...................................................166
A.43 StateConstructW.h...........................................169
A.44 StateConstructW.c...........................................170
A.45 StateSearchW.h..............................................171
A.46 StateSearchW.c..............................................172
A.47 syntFilter.h................................................176
A.48 syntFilter.c................................................176
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1. INTRODUCTION
This document contains the description of an algorithm for the
coding of speech signals sampled at 8 kHz. The algorithm, called
iLBC, uses a block-independent linear-predictive coding (LPC)
algorithm and has support for two basic frame lengths: 20 ms at 15.2
kbit/s and 30 ms at 13.33 kbit/s. When the codec operates at block
lengths of 20 ms, it produces 304 bits per block which SHOULD be
packetized as in [1]. Similarly, for block lengths of 30 ms it
produces 400 bits per block which SHOULD be packetized as in [1].
The two modes for the different frame sizes operate in a very
similar way. When they differ it is explicitly stated in the text,
usually with the notation x/y, where x refers to the 20 ms mode and
y refers to the 30 ms mode.
The described algorithm results in a speech coding system with a
controlled response to packet losses similar to what is known from
pulse code modulation (PCM) with packet loss concealment (PLC), such
as the ITU-T G.711 standard [4] which operates at a fixed bit rate
of 64 kbit/s. At the same time, the described algorithm enables
fixed bit rate coding with a quality-versus-bit rate tradeoff close
to state-of-the-art. A suitable RTP payload format for the iLBC
codec is specified in [1].
Some of the applications for which this coder is suitable are: real
time communications such as telephony and videoconferencing,
streaming audio, archival, and messaging.
Cable Television Laboratories (CableLabs(R)) intends to adapt iLBC
as a PacketCable(TM) audio codec standard for VoIP over Cable
applications [3].
This document is organized as follows. In Section 2 a brief outline
of the codec is given. The specific encoder and decoder algorithms
are explained in Sections 3 and 4, respectively. A c-code reference
implementation is provided in Appendix A.
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 [2].
2. OUTLINE OF THE CODEC
The codec consists of an encoder and a decoder described in Section
2.1 and 2.2, respectively.
The essence of the codec is LPC and block based coding of the LPC
residual signal. For each 160/240 (20ms/30 ms) sample block, the
following major steps are performed: A set of LPC filters are
computed and the speech signal is filtered through them to produce
the residual signal. The codec uses scalar quantization of the
dominant part, in terms of energy, of the residual signal for the
block. The dominant state is of length 57/58 (20 ms/30 ms) samples
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and forms a start state for dynamic codebooks constructed from the
already coded parts of the residual signal. These dynamic codebooks
are used to code the remaining parts of the residual signal. By this
method, coding independence between blocks is achieved, resulting in
elimination of propagation of perceptual degradations due to packet
loss. The method facilitates high-quality packet loss concealment
(PLC).
2.1 Encoder
The input to the encoder SHOULD be 16 bit uniform PCM sampled at 8
kHz. It SHOULD be partitioned into blocks of BLOCKL=160/240 samples
for the 20/30 ms frame size. Each block is divided into NSUB=4/6
consecutive sub-blocks of SUBL=40 samples each. For 30 ms frame
size, the encoder performs two LPC_FILTERORDER=10 linear-predictive
coding (LPC) analyses. The first analysis applies a smooth window
centered over the 2nd sub-block and extending to the middle of the
5th sub-block. The second LPC analysis applies a smooth asymmetric
window centered over the 5th sub-block and extending to the end of
the 6th sub-block. For 20 ms frame size one LPC_FILTERORDER=10
linear-predictive coding (LPC) analysis is performed with a smooth
window centered over the 3rd sub-frame.
For each of the LPC analyses, a set of line-spectral frequencies
(LSFs) are obtained, quantized and interpolated to obtain LSF
coefficients for each sub-block. Subsequently, the LPC residual is
computed using the quantized and interpolated LPC analysis filters.
The two consecutive sub-blocks of the residual exhibiting the
maximal weighted energy are identified. Within these 2 sub-blocks,
the start state (segment) is selected from two choices: the first
57/58 samples or the last 57/58 samples of the 2 consecutive sub-
blocks. The selected segment is the one of higher energy. The start
state is encoded with scalar quantization.
A dynamic codebook encoding procedure is used to encode 1) the 23/22
(20 ms/30 ms) remaining samples in the 2 sub-blocks containing the
start state; 2) encoding of the sub-blocks after the start state in
time; 3) encoding of the sub-blocks before the start state in time.
Thus, the encoding target can be either the 23/22 samples remaining
of the 2 sub-blocks containing the start state or a 40 sample sub-
block. This target can consist of samples that are indexed forwards
in time or backwards in time depending on the location of the start
state.
The coding is based on an adaptive codebook that is built from a
codebook memory which contains decoded LPC excitation samples from
the already encoded part of the block. These samples are indexed in
the same time direction as the target vector and ending at the
sample instant prior to the first sample instant represented in the
target vector. The codebook is used in CB_NSTAGES=3 stages in a
successive refinement approach and the resulting 3 code vector gains
are encoded with 5, 4, and 3 bit scalar quantization, respectively.
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The codebook search method employs noise shaping derived from the
LPC filters and the main decision criterion is minimizing the
squared error between the target vector and the code vectors. Each
code vector in this codebook comes from one of CB_EXPAND=2 codebook
sections. The first section is filled with delayed, already encoded
residual vectors. The code vectors of the second codebook section
are constructed by predefined linear combinations of vectors in the
first section of the codebook.
Since codebook encoding with squared-error matching is known to
produce a coded signal of less power than the scalar quantized start
state signal, a gain re-scaling method is implemented by a refined
search for a better set of codebook gains in terms of power matching
after encoding. This is done by searching for a higher value of the
gain factor for the first stage codebook since the subsequent stage
codebook gains are scaled by the first stage gain.
2.2 Decoder
For packet communications, typically a jitter buffer placed at the
receiving end decides whether the packet containing an encoded
signal block has been received or lost. This logic is not part of
the codec described here. For each received encoded signal block the
decoder performs a decoding. For each lost signal block the decoder
performs a PLC operation.
The decoding for each block starts by decoding and interpolating the
LPC coefficients. Subsequently the start state is decoded.
For codebook encoded segments, each segment is decoded by
constructing the 3 code vectors given by the received codebook
indices in the same way as the code vectors were constructed in the
encoder. The 3 gain factors are also decoded and the resulting
decoded signal is given by the sum of the 3 codebook vectors scaled
with respective gain.
An enhancement algorithm is applied on the reconstructed excitation
signal. This enhancement augments the periodicity of voiced speech
regions. The enhancement is optimized under the constraint that the
modification signal (defined as the difference between the enhanced
excitation and the excitation signal prior to enhancement) has a
short-time energy that does not exceed a preset fraction of the
short-time energy of the excitation signal prior to enhancement.
A packet loss concealment (PLC) operation is easily embedded in the
decoder. The PLC operation can, e.g., be based on repetition of LPC
filters and obtaining the LPC residual signal using a long term
prediction estimate from previous residual blocks.
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3. ENCODER PRINCIPLES
The following block diagram is an overview of all the components of
the iLBC encoding procedure. The description of the blocks contains
references to the section where that particular procedure is
described further.
+-----------+ +---------+ +---------+
speech -> | 1. Pre P | -> | 2. LPC | -> | 3. Ana | ->
+-----------+ +---------+ +---------+
+---------------+ +--------------+
-> | 4. Start Sel | ->| 5. Scalar Qu | ->
+---------------+ +--------------+
+--------------+ +---------------+
-> |6. CB Search | -> | 7. Packetize | -> payload
| +--------------+ | +---------------+
----<---------<------
sub-frame 0..2/4 (20 ms/30 ms)
Figure 3.1. Flow chart of the iLBC encoder
1. Pre process speech with a HP filter if needed (section 3.1)
2. Compute LPC parameters, quantize and interpolate (section 3.2)
3. Use analysis filters on speech to compute residual (section 3.3)
4. Select position of 57/58 sample start state (section 3.5)
5. Quantize the 57/58 sample start state with scalar quantization
(section 3.5)
6. Search the codebook for each sub-frame. Start with 23/22 sample
block, then encode sub-blocks forward in time and then encode sub-
blocks backward in time. For each block the steps in figure 3.4 are
performed (section 3.6)
7. Packetize the bits into the payload specified in table 3.2.
The input to the encoder SHOULD be 16 bit uniform PCM sampled at 8
kHz. Also it SHOULD be partitioned into blocks of BLOCKL=160/240
samples. Each block input to the encoder is divided into NSUB=4/6
consecutive sub-blocks of SUBL=40 samples each.
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0 39 79 119 159
+---------------------------------------+
| 1 | 2 | 3 | 4 |
+---------------------------------------+
20 ms frame
0 39 79 119 159 199 239
+-----------------------------------------------------------+
| 1 | 2 | 3 | 4 | 5 | 6 |
+-----------------------------------------------------------+
30 ms frame
Figure 3.2. One input block to the encoder for 20 ms (with 4 sub-
frames) and 30 ms (with 6 sub-frames).
3.1 Pre-processing
In some applications the recorded speech signal contains DC level
and/or 50/60 Hz noise. If these components have not been removed
prior to the encoder call, they should be removed by a high-pass
filter. A reference implementation of this, using a filter with cut
off frequency 90 Hz, can be found in Appendix A.28.
3.2 LPC Analysis and Quantization
The input to the LPC analysis module is a possibly high-pass
filtered speech buffer, speech_hp, that contains 240/300
(LPC_LOOKBACK + BLOCKL = 80/60 + 160/240 = 240/300) speech samples,
where samples 0 through 79/59 are from the previous block and
samples 80/60 through 239/299 are from the current block. No look-
ahead into the next block is used. For the very first block
processed, the look back samples are assumed to be zeros.
For each input block, the LPC analysis calculates one/two set(s) of
LPC_FILTERORDER=10 LPC filter coefficients using the autocorrelation
method and the Levinson-Durbin recursion. These coefficients are
converted to the Line Spectrum Frequency representation. In the 20
ms case the set, lsf, represents the spectral characteristics as
measured at the center of the third sub-block. For 30 ms frames the
first set, lsf1, represents the spectral properties of the input
signal at the center of the second sub-block while the other set,
lsf2, represents the spectral characteristics as measured at the
center of the fifth sub-block. The details of the computation for 30
ms frames are described in 3.2.1 through 3.2.6. Section 3.2.7
explains how the LPC Analysis and Quantization differs for 20 ms
frames.
3.2.1 Computation of Autocorrelation Coefficients
The first step in the LPC analysis procedure is to calculate
autocorrelation coefficients using windowed speech samples. This
windowing is the only difference in the LPC analysis procedure for
the two sets of coefficients. For the first set, a 240 sample long
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standard symmetric Hanning window is applied to samples 0 through
239 of the input data. The first window, lpc_winTbl, is defined as:
lpc_winTbl[i]= 0.5 * (1.0 - cos((2*PI*(i+1))/(BLOCKL+1)));
i=0,...,119
lpc_winTbl[i] = winTbl[BLOCKL - i - 1]; i=120,...,239
The windowed speech speech_hp_win1 is then obtained by multiplying
the 240 first samples of the input speech buffer with the window
coefficients:
speech_hp_win1[i] = speech_hp[i] * lpc_winTbl[i];
i=0,...,BLOCKL-1
From these 240 windowed speech samples, 11 (LPC_FILTERORDER + 1)
autocorrelation coefficients, acf1, are calculated:
acf1[lag] += speech_hp_win1[n] * speech_hp_win1[n + lag];
lag=0,...,LPC_FILTERORDER; n=0,...,BLOCKL-lag-1
In order to make the analysis more robust against numerical
precision problems, a spectral smoothing procedure is applied by
windowing the autocorrelation coefficients before the LPC
coefficients are computed. Also, a white noise floor is added to the
autocorrelation function by multiplying coefficient zero by 1.0001
(40dB below the energy of the windowed speech signal). These two
steps are implemented by multiplying the autocorrelation
coefficients with the following window:
lpc_lagwinTbl[0] = 1.0001;
lpc_lagwinTbl[i] = exp(-0.5 * ((2 * PI * 60.0 * i) /FS)^2);
i=1,...,LPC_FILTERORDER
where FS=8000 is the sampling frequency
Then, the windowed acf function acf1_win is obtained by:
acf1_win[i] = acf1[i] * lpc_lagwinTbl[i];
i=0,...,LPC_FILTERORDER
The second set of autocorrelation coefficients, acf2_win are
obtained in a similar manner. The window, lpc_asymwinTbl, is applied
to samples 60 through 299, i.e., the entire current block. The
window consists of two segments; the first (samples 0 to 219) being
half a Hanning window with length 440 and the second being a quarter
of a cycle of a cosine wave. By using this asymmetric window, an LPC
analysis centered in the fifth sub-block is obtained without the
need for any look-ahead, which would have added delay. The
asymmetric window is defined as:
lpc_asymwinTbl[i] = (sin(PI * (i + 1) / 441))^2; i=0,...,219
lpc_asymwinTbl[i] = cos((i - 220) * PI / 40); i=220,...,239
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and the windowed speech is computed by:
speech_hp_win2[i] = speech_hp[i + LPC_LOOKBACK] *
lpc_asymwinTbl[i]; i=0,....BLOCKL-1
The windowed autocorrelation coefficients are then obtained in
exactly the same way as for the first analysis instance.
The generation of the windows lpc_winTbl, lpc_asymwinTbl, and
lpc_lagwinTbl are typically done in advance and the arrays are
stored in ROM rather than repeating the calculation for every block.
3.2.2 Computation of LPC Coefficients
From the 2 x 11 smoothed autocorrelation coefficients, acf1_win and
acf2_win, the 2 x 11 LPC coefficients, lp1 and lp2, are calculated
in the same way for both analysis locations using the well known
Levinson-Durbin recursion. The first LPC coefficient is always 1.0,
resulting in 10 unique coefficients.
After determining the LPC coefficients, a bandwidth expansion
procedure is applied in order to smooth the spectral peaks in the
short-term spectrum. The bandwidth addition is obtained by the
following modification of the LPC coefficients:
lp1_bw[i] = lp1[i] * chirp^i; i=0,...,LPC_FILTERORDER
lp2_bw[i] = lp2[i] * chirp^i; i=0,...,LPC_FILTERORDER
where "chirp" is a real number between 0 and 1. It is RECOMMENDED to
use a value of 0.9.
3.2.3 Computation of LSF Coefficients from LPC Coefficients
Thusfar, two sets of LPC coefficients that represent the short-term
spectral characteristics of the speech signal for two different time
locations within the current block have been determined. These
coefficients SHOULD be quantized and interpolated. Before doing so,
it is advantageous to convert the LPC parameters into another type
of representation called Line Spectral Frequencies (LSF). The LSF
parameters are used because they are better suited for quantization
and interpolation than the regular LPC coefficients. Many
computationally efficient methods for calculating the LSFs from the
LPC coefficients have been proposed in the literature. The detailed
implementation of one applicable method can be found in Appendix
A.26. The two arrays of LSF coefficients obtained, lsf1 and lsf2,
are of dimension 10 (LPC_FILTERORDER).
3.2.4 Quantization of LSF Coefficients
Since the LPC filters defined by the two sets of LSFs are needed
also in the decoder, the LSF parameters need to be quantized and
transmitted as side information. The total number of bits required
to represent the quantization of the two LSF representations for one
block of speech is 40 with 20 bits used for each of lsf1 and lsf2.
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For computational and storage reasons, the LSF vectors are quantized
using 3-split vector quantization (VQ). That is, the LSF vectors are
split into three sub-vectors which are each quantized with a regular
VQ. The quantized versions of lsf1 and lsf2, qlsf1 and qlsf2, are
obtained by using the same memoryless split VQ. The length of each
of these two LSF vectors is 10 and they are split into 3 sub-vectors
containing 3, 3 and 4 values respectively.
For each of the sub-vectors, a separate codebook of quantized values
has been designed using a standard VQ training method for a large
database containing speech from a large number of speakers recorded
under various conditions. The size of each of the three codebooks
associated with the split definitions above is:
int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
The actual values of the vector quantization codebook that must be
used can be found in the reference code of appendix A. Both sets of
LSF coefficients, lsf1 and lsf2, are quantized with a standard
memoryless split vector quantization (VQ) structure using the
squared error criterion in the LSF domain. The split VQ quantization
consists of the following steps:
1) Quantize the first 3 LSF coefficients (1 - 3) with a VQ codebook
of size 64.
2) Quantize the LSF coefficients 4, 5, and 6 with VQ a codebook of
size 128.
3) Quantize the last 4 LSF coefficients (7 - 10) with a VQ codebook
of size 128.
This procedure, repeated for lsf1 and lsf2, gives 6 quantization
indices and the quantized sets of LSF coefficients qlsf1 and qlsf2.
Each set of three indices is encoded with 6 + 7 + 7 = 20 bits. The
total number of bits used for LSF quantization in a block is thus 40
bits.
3.2.5 Stability Check of LSF Coefficients
The LSF representation of the LPC filter has the nice property that
the coefficients are ordered by increasing value, i.e., lsf(n-1) <
lsf(n), 0 < n < 10, if the corresponding synthesis filter is stable.
Since we are employing a split VQ scheme it is possible that at the
split boundaries the LSF coefficients are not ordered correctly and
hence the corresponding LP filter is unstable. To ensure that the
filter used is stable, a stability check is performed for the
quantized LSF vectors. If it turns out that the coefficients are not
ordered appropriately (with a safety margin of 50 Hz to ensure that
formant peaks are not too narrow) they will be moved apart. The
detailed method for this can be found in Appendix A.40. The same
procedure is performed in the decoder. This ensures that exactly the
same LSF representations are used in both encoder and decoder.
3.2.6 Interpolation of LSF Coefficients
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From the two sets of LSF coefficients that are computed for each
block of speech, different LSFs are obtained for each sub-block by
means of interpolation. This procedure is performed for the original
LSFs (lsf1 and lsf2), as well as the quantized versions qlsf1 and
qlsf2 since both versions are used in the encoder. Here follows a
brief summary of the interpolation scheme while the details are
found in the c-code of Appendix A. In the first sub-block, the
average of the second LSF vector from the previous block and the
first LSF vector in the current block is used. For sub-blocks two
through five the LSFs used are obtained by linear interpolation from
lsf1 (and qlsf1) to lsf2 (and qlsf2) with lsf1 used in sub-block two
and lsf2 in sub-block five. In the last sub-block, lsf2 is used. For
the very first block it is assumed that the last LSF vector of the
previous block is equal to a predefined vector, lsfmeanTbl, that was
obtained by calculating the mean LSF vector of the LSF design
database.
lsfmeanTbl[LPC_FILTERORDER] = {0.281738, 0.445801, 0.663330,
0.962524, 1.251831, 1.533081, 1.850586, 2.137817,
2.481445, 2.777344}
The interpolation method is standard linear interpolation in the LSF
domain. The interpolated LSF values are converted to LPC
coefficients for each sub-block. The unquantized and quantized LPC
coefficients form two sets of filters respectively. The unquantized
analysis filter for sub-block k:
___
\
Ak(z)= 1 + > ak(i)*z^(-i)
/__
i=1...LPC_FILTERORDER
And the quantized analysis filter for sub-block k:
___
\
A~k(z)= 1 + > a~k(i)*z^(-i)
/__
i=1...LPC_FILTERORDER
A reference implementation of the lsf encoding is given in Appendix
A.38. A reference implementation of the corresponding decoding can
be found in Appendix A.36.
3.2.7 LPC Analysis and Quantization for 20 ms frames
As stated before, the codec only calculates one set of LPC
parameters for the 20 ms frame size as opposed to two sets for 30 ms
frames. A single set of autocorrelation coefficients is calculated
on the LPC_LOOKBACK + BLOCKL = 80 + 160 = 240 samples. These samples
are windowed with the asymmetric window lpc_asymwinTbl, centered
over the third sub-frame, to form speech_hp_win. Autocorrelation
coefficients, acf, are calculated on the 240 samples in
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speech_hp_win and then windowed exactly as in 3.2.1 (resulting in
acf_win).
This single set of windowed autocorrelation coefficients is used to
calculate LPC Coefficients, LSF Coefficients and quantized LSF
coefficients in exactly the same manner as in 3.2.3 to 3.2.4. As for
the 30 ms frame size, the 10 LSF coefficients are divided into three
sub-vectors of size 3, 3, 4 and quantized using the same scheme and
codebook as in 3.2.4 to finally get 3 quantization indices. The
quantized LSF coefficients are stabilized with the algorithm
described in 3.2.5.
From the set of LSF coefficients that was computed for this block
together with the LSF coefficients from the previous block,
different LSFs are obtained for each sub-block by means of
interpolation. The interpolation is done linearly in the LSF domain
over the 4 sub-blocks, so that the n-th sub-frame uses the weight
(4-n)/4 for the LSF from old frame and the weight n/4 of the LSF
from the current frame. For the very first block the mean LSF,
lsfmeanTbl, is used as the LSF from the previous block. Similar to
3.2.6, both unquantized, A(z), and quantized, A~(z), analysis
filters are calculated for each of the four sub-blocks.
3.3 Calculation of the Residual
The block of speech samples is filtered by the quantized and
interpolated LPC analysis filters to yield the residual signal. In
particular, the corresponding LPC analysis filter for each 40 sample
sub-block is used to filter the speech samples for the same sub-
block. The filter memory at the end of each sub-block is carried
over to the LPC filter of the next sub-block. The signal at the
output of each LP analysis filter constitutes the residual signal
for the corresponding sub-block.
A reference implementation of the LPC analysis filters is given in
Appendix A.10.
3.4 Perceptual Weighting Filter
In principle any good design of a perceptual weighting filter can be
applied in the encoder without compromising this codec definition.
It is however RECOMMENDED to use the perceptual weighting filter
specified below:
Weighting filter for sub-block k:
Wk(z)=1/Ak(z/LPC_CHIRP_WEIGHTDENUM), where
LPC_CHIRP_WEIGHTDENUM = 0.4222
This is a simple design with low complexity that is applied in the
LPC residual domain. Here Ak(z) is the filter obtained from
unquantized but interpolated LSF coefficients.
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3.5 Start State Encoder
The start state is quantized using a common 6-bit scalar quantizer
for the block and a 3-bit scalar quantizer operating on scaled
samples in the weighted speech domain. In the following we describe
the state encoding in greater detail.
3.5.1 Start State Estimation
The two sub-blocks containing the start state are determined by
finding the two consecutive sub-blocks in the block having the
highest power. Advantageously, down-weighting is used in the
beginning and end of the sub-frames. I.e., the following measure is
computed (NSUB=4/6 for 20/30 ms frame size):
nsub=1,...,NSUB-1
ssqn[nsub] = 0.0;
for (i=(nsub-1)*SUBL; i<(nsub-1)*SUBL+5; i++)
ssqn[nsub] += sampEn_win[i-(nsub-1)*SUBL]*
residual[i]*residual[i];
for (i=(nsub-1)*SUBL+5; i<(nsub+1)*SUBL-5; i++)
ssqn[nsub] += residual[i]*residual[i];
for (i=(nsub+1)*SUBL-5; i<(nsub+1)*SUBL; i++)
ssqn[nsub] += sampEn_win[(nsub+1)*SUBL-i-1]*
residual[i]*residual[i];
where sampEn_win[5]={1/6, 2/6, 3/6, 4/6, 5/6}; MAY be used. The sub-
frame number corresponding to the maximum value of ssqEn_win[nsub-
1]*ssqn[nsub] is selected as the start state indicator. A weighting
of ssqEn_win[]={0.8,0.9,1.0,0.9,0.8} for 30 ms frames and
ssqEn_win[]={0.9,1.0,0.9} for 20 ms frames; MAY advantageously be
used to bias the start state towards the middle of the frame.
For 20 ms frames there are 3 possible positions of the two-sub-block
length maximum power segment, the start state position is encoded
using 2 bits. The start state position, start, MUST be encoded as:
start=1: start state in sub-frame 0 and 1
start=2: start state in sub-frame 1 and 2
start=3: start state in sub-frame 2 and 3
For 30 ms frames there are 5 possible positions of the two-sub-block
length maximum power segment, the start state position is encoded
using 3 bits. The start state position, start, MUST be encoded as:
start=1: start state in sub-frame 0 and 1
start=2: start state in sub-frame 1 and 2
start=3: start state in sub-frame 2 and 3
start=4: start state in sub-frame 3 and 4
start=5: start state in sub-frame 4 and 5
hence, in both cases, index 0 is not utilized. In order to shorten
the start state for bit rate efficiency, the start state is brought
down to STATE_SHORT_LEN=57 samples for 20 ms frames and
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STATE_SHORT_LEN=58 samples for 30 ms frames. The power of the first
23/22 and last 23/22 samples of the 2 sub-frame block identified
above is computed as the sum of the squared signal sample values and
the 23/22 sample segment with the lowest power is excluded from the
start state. One bit is transmitted to indicate which of the 2
possible 57/58 sample segments is used. The start state position
within the 2 sub-frames determined above, state_first, MUST be
encoded as:
state_first=1: start state is first STATE_SHORT_LEN samples
state_first=0: start state is last STATE_SHORT_LEN samples
3.5.2 All-Pass Filtering and Scale Quantization
The block of residual samples in the start state is first filtered
by an all-pass filter with the quantized LPC coefficients as
denominator and reversed quantized LPC coefficients as numerator.
The purpose of this phase-dispersion filter is to get a more even
distribution of the sample values in the residual signal. The
filtering is performed by circular convolution, where the initial
filter memory is set to zero.
res(0..(STATE_SHORT_LEN-1)) = uncoded start state residual
res((STATE_SHORT_LEN)..(2*STATE_SHORT_LEN-1)) = 0
Pk(z) = A~rk(z)/A~k(z), where
___
\
A~rk(z)= z^(-LPC_FILTERORDER)+>a~k(i+1)*z^(i-(LPC_FILTERORDER-1))
/__
i=0...(LPC_FILTERORDER-1)
and A~k(z) is taken from the block where the start state begins
res -> Pk(z) -> filtered
ccres(k) = filtered(k) + filtered(k+STATE_SHORT_LEN),
k=0..(STATE_SHORT_LEN-1)
The all pass filtered block is searched for its largest magnitude
sample. The 10-logarithm of this magnitude is quantized with a 6-bit
quantizer, state_frgqTbl, by finding the nearest representation.
This results in an index, idxForMax, corresponding to a quantized
value, qmax. The all-pass filtered residual samples in the block are
then multiplied with a scaling factor scal=4.5/(10^qmax) to yield
normalized samples.
state_frgqTbl[64] = {1.000085, 1.071695, 1.140395, 1.206868,
1.277188, 1.351503, 1.429380, 1.500727, 1.569049,
1.639599, 1.707071, 1.781531, 1.840799, 1.901550,
1.956695, 2.006750, 2.055474, 2.102787, 2.142819,
2.183592, 2.217962, 2.257177, 2.295739, 2.332967,
2.369248, 2.402792, 2.435080, 2.468598, 2.503394,
2.539284, 2.572944, 2.605036, 2.636331, 2.668939,
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2.698780, 2.729101, 2.759786, 2.789834, 2.818679,
2.848074, 2.877470, 2.906899, 2.936655, 2.967804,
3.000115, 3.033367, 3.066355, 3.104231, 3.141499,
3.183012, 3.222952, 3.265433, 3.308441, 3.350823,
3.395275, 3.442793, 3.490801, 3.542514, 3.604064,
3.666050, 3.740994, 3.830749, 3.938770, 4.101764}
3.5.3 Scalar Quantization
The normalized samples are quantized in the perceptually weighted
speech domain by a sample-by-sample scalar DPCM quantization as
depicted in Figure 3.3. Each sample in the block is filtered by a
weighting filter Wk(z), specified in section 3.4, to form a weighted
speech sample x[n]. The target sample d[n] is formed by subtracting
a predicted sample y[n], where the prediction filter is given by
Pk(z) = 1 - 1 / Wk(z).
+-------+ x[n] + d[n] +-----------+ u[n]
residual -->| Wk(z) |-------->(+)---->| Quantizer |------> quantized
+-------+ - /|\ +-----------+ | residual
| \|/
y[n] +--------------------->(+)
| |
| +------+ |
+--------| Pk(z)|<------+
+------+
Figure 3.3. Quantization of start state samples by DPCM in weighted
speech domain.
The coded state sample u[n] is obtained by quantizing d[n] with a 3-
bit quantizer with quantization table state_sq3Tbl.
state_sq3Tbl[8] = {-3.719849, -2.177490, -1.130005, -0.309692,
0.444214, 1.329712, 2.436279, 3.983887}
The quantized samples are transformed back to the residual domain by
1) scaling with 1/scal 2) time-reversing the scaled samples 3)
filtering the time-reversed samples by the same all-pass filter as
in section 3.5.2, using circular convolution 4) time-reversing the
filtered samples. (More detailed in section 4.2)
A reference implementation of the start state encoding can be found
in Appendix A.46.
3.6 Encoding the remaining samples
A dynamic codebook is used to encode 1) the 23/22 remaining samples
in the 2 sub-blocks containing the start state; 2) encoding of the
sub-blocks after the start state in time; 3) encoding of the sub-
blocks before the start state in time. Thus, the encoding target can
be either the 23/22 samples remaining of the 2 sub-blocks containing
the start state or a 40 sample sub-block. This target can consist of
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samples that are indexed forwards in time or backwards in time
depending on the location of the start state. The length of the
target is denoted by lTarget.
The coding is based on an adaptive codebook that is built from a
codebook memory which contains decoded LPC excitation samples from
the already encoded part of the block. These samples are indexed in
the same time direction as the target vector and ending at the
sample instant prior to the first sample instant represented in the
target vector. The codebook memory has length lMem which is equal to
CB_MEML=147 for the two/four 40 sample sub-blocks and 85 for the
23/22 sample sub-block.
The following figure shows an overview of the encoding procedure.
+------------+ +---------------+ +-------------+
-> | 1. Decode | -> | 2. Mem setup | -> | 3. Perc. W. | ->
+------------+ +---------------+ +-------------+
+------------+ +-----------------+
-> | 4. Search | -> | 5. Upd. Target | ------------------>
| +------------+ +------------------ |
----<-------------<-----------<----------
stage=0..2
+----------------+
-> | 6. Recalc G[0] | ---------------> gains and CB indices
+----------------+
Figure 3.4. Flow chart of the codebook search in the iLBC encoder
1. Decode the part of the residual that has been encoded so far,
using the codebook without perceptual weighting
2. Set up the memory by taking data from the decoded residual. This
memory is used to construct codebooks from. For blocks preceding the
start state, both the decoded residual and the target are time
reversed (section 3.6.1)
3. Filter the memory + target with the perceptual weighting filter
(section 3.6.2)
4. Search for the best match between the target and the codebook
vector. Compute the optimal gain for this match and quantize that
gain (section 3.6.4)
5. Update the perceptually weighted target by subtracting the
contribution from the selected codebook vector from the perceptually
weighted memory (quantized gain times selected vector). Repeat 4.
and 5. for the 2 additional stages
6. Calculate the energy loss due to encoding of the residual. If
needed, compensate for this loss by an upscaling and requantization
of the gain for the first stage (section 3.7)
The following sections provide an in-depth description of the
different blocks of figure 3.4.
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3.6.1 Codebook Memory
The codebook memory is based on the already encoded sub-blocks so
the available data for encoding increases for each new sub-block
that has been encoded. Until enough sub-blocks have been encoded to
fill the codebook memory with data it is padded with zeros. The
following figure shows an example of the order in which the sub-
blocks are encoded for the 30 ms frame size if the start state is
located in the last 58 samples of sub-block 2 and 3.
+-----------------------------------------------------+
| 5 | 1 |///|////////| 2 | 3 | 4 |
+-----------------------------------------------------+
Figure 3.5. The order from 1 to 5 in which the sub-blocks are
encoded. The slashed area is the start state.
The first target sub-block to be encoded is number 1 and the
corresponding codebook memory is shown in the following figure.
Since the target vector is before the start state in time the
codebook memory and target vector are time reversed. By reversing
them in time, the search algorithm can be reused. Since only the
start state has been encoded so far the last samples of the codebook
memory are padded with zeros.
+-------------------------
|zeros|\\\\\\\\|\\\\| 1 |
+-------------------------
Figure 3.6. The codebook memory, length lMem=85 samples, and the
target vector 1, length 22 samples.
The next step is to encode sub-block 2 using the memory which now
has increased since sub-block 1 has been encoded. The following
figure shows the codebook memory for encoding of sub-block 2.
+-----------------------------------
| zeros | 1 |///|////////| 2 |
+-----------------------------------
Figure 3.7. The codebook memory, length lMem=147 samples, and the
target vector 2, length 40 samples.
The next step is to encode sub-block 3 using the memory which now
has increased yet again since sub-blocks 1 and 2 have been encoded
but it still has to be padded with a few zeros. The following figure
shows the codebook memory for encoding of sub-block 3.
+------------------------------------------
|zeros| 1 |///|////////| 2 | 3 |
+------------------------------------------
Figure 3.8. The codebook memory, length lMem=147 samples, and the
target vector 3, length 40 samples.
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The next step is to encode sub-block 4 using the memory which now
has increased yet again since sub-blocks 1, 2 and 3 have been
encoded. This time the memory does not have to be padded with zeros.
The following figure shows the codebook memory for encoding of sub-
block 4.
+------------------------------------------
|1|///|////////| 2 | 3 | 4 |
+------------------------------------------
Figure 3.9. The codebook memory, length lMem=147 samples, and the
target vector 4, length 40 samples.
The final target sub-block to be encoded is number 5 and the
corresponding codebook memory is shown in the following figure.
Since the target vector is before the start state in time the
codebook memory and target vector are time reversed.
+-------------------------------------------
| 3 | 2 |\\\\\\\\|\\\\| 1 | 5 |
+-------------------------------------------
Figure 3.10. The codebook memory, length lMem=147 samples, and the
target vector 5, length 40 samples.
For the case of 20 ms frames the encoding procedure looks almost
exactly the same. The only difference is that the size of the start
state is 57 samples and that there are only 3 sub-blocks to be
encoded. The encoding order is the same as above starting with the
23 sample target and then encoding the two remaining 40 sample sub-
blocks, first going forward in time and then going backwards in time
relative to the start state.
3.6.2 Perceptual Weighting of Codebook Memory and Target
To provide a perceptual weighting of the coding error, a
concatenation of the codebook memory and the target to be coded is
all pole filtered with the perceptual weighting filter specified in
section 3.4. The filter state of the weighting filter is set to
zero.
in(0..(lMem-1)) = unweighted codebook memory
in(lMem..(lMem+lTarget-1)) = unweighted target signal
in -> Wk(z) -> filtered,
where Wk(z) is taken from the sub-block of the target
weighted codebook memory = filtered(0..(lMem-1))
weighted target signal = filtered(lMem..(lMem+lTarget-1))
The codebook search is done using the weighted codebook memory and
the weighted target, while the decoding and the codebook memory
update uses the unweighted codebook memory.
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3.6.3 Codebook Creation
The codebook for the search is created from the perceptually
weighted codebook memory. It consists of two sections where the
first is referred to as the base codebook and the second as the
expanded codebook since it is created by linear combinations of the
first. Each of these two sections also has a subsection referred to
as the augmented codebook. The augmented codebook is only created
and used for the coding of the 40 sample sub-blocks and not for the
23/22 sample sub-block case. The codebook size used for the
different sub-blocks and different stages are summarized in the
table below.
Stage
1 2 & 3
--------------------------------------------
22 128 (64+0)*2 128 (64+0)*2
Sub- 1:st 40 256 (108+20)*2 128 (44+20)*2
Blocks 2:nd 40 256 (108+20)*2 256 (108+20)*2
3:rd 40 256 (108+20)*2 256 (108+20)*2
4:th 40 256 (108+20)*2 256 (108+20)*2
Table 3.1. Codebook sizes for the 30 ms mode
The table 3.1 shows the codebook size for the different sub-blocks
and stages for 30 ms frames. Inside the parenthesis it shows how the
number of codebook vectors is distributed, within the two sections,
between the base/expanded codebook and the augmented base/expanded
codebook. It should be interpreted in the following way:
(base/expanded cb + augmented base/expanded cb). The total number of
codebook vectors for a specific sub-block and stage is given by the
following formula:
Tot. cb vectors = base cb + aug. base cb + exp. cb + aug. exp. cb
The corresponding values to figure 3.1 for 20 ms frames are only
slightly modified. The short sub-block is 23 instead of 22 samples
and the 3:rd and 4:th sub-frame are not present.
3.6.3.1 Creation of a Base Codebook
The base codebook is given by the perceptually weighted codebook
memory that is mentioned in section 3.5.3. The different codebook
vectors are given by sliding a window of length 23/22 or 40, given
by variable lTarget, over the lMem long perceptually weighted
codebook memory. The indices are ordered so that the codebook vector
containing sample(lMem-lTarget-n) to (lMem-n-1) of the codebook
memory vector has index n, where n=0..lMem-lTarget. Thus the total
number of base codebook vectors is lMem-lTarget+1 and the indices
are ordered from sample delay lTarget (23/22 or 40) to lMem+1 (86 or
148).
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3.6.3.2 Codebook Expansion
The base codebook is expanded by a factor of 2, creating an
additional section in the codebook. This new section is obtained by
filtering the base codebook, base_cb, with a FIR filter with filter
length CB_FILTERLEN=8. The delay of four samples introduced by the
FIR filter is compensated for in the construction of the expanded
codebook.
cbfiltersTbl[CB_FILTERLEN]={-0.033691, 0.083740, -0.144043,
0.713379, 0.806152, -0.184326,
0.108887, -0.034180};
___
\
exp_cb(k)= + > cbfiltersTbl(i)*x(k-i+4)
/__
i=0...(LPC_FILTERORDER-1)
where x(j) = base_cb(j) for j=0..lMem-1 and 0 otherwise
The individual codebook vectors of the new filtered codebook,
exp_cb, and their indices are obtained in the same fashion as
described above for the base codebook.
3.6.3.3 Codebook Augmentation
For the cases when encoding entire sub-blocks, i.e. cbveclen=40, the
base and expanded codebooks are augmented to increase codebook
richness. The codebooks are augmented by vectors produced by
interpolation of segments. The base and expanded codebook,
constructed above, consists of vectors corresponding to sample
delays in the range from cbveclen to lMem. The codebook augmentation
attempts to augment these codebooks with vectors corresponding to
sample delays from 20 to 39. However, not all of these samples are
present in the base codebook and expanded codebook respectively.
Therefore, the augmentation vectors are constructed as linear
combinations between samples corresponding to sample delays in the
range 20 to 39. The general idea of this procedure is presented in
the following figures and text. The procedure is performed for both
the base codebook and the expanded codebook.
- - ------------------------|
codebook memory |
- - ------------------------|
|-5-|---15---|-5-|
pi pp po
| | Codebook vector
|---15---|-5-|-----20-----| <- corresponding to
i ii iii sample delay 20
Figure 3.11. Generation of the first augmented codebook
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The figure 3.11 shows the codebook memory with pointers pi, pp and
po where pi points to sample 25, pp to sample 20 and po to sample 5.
Below the codebook memory, the augmented codebook vector
corresponding to sample delay 20 is drawn. Segment i consists of 15
samples from pointer pp and forward in time. Segment ii consists of
5 interpolated samples from pi and forward and from po and forward.
The samples are linearly interpolated with weights [0.0, 0.2, 0.4,
0.6, 0.8] for pi and weights [1.0, 0.8, 0.6, 0.4, 0.2] for po.
Segment iii consists of 20 samples from pp and forward. The
augmented codebook vector corresponding to sample delay 21 is
produced by moving pointers pp and pi one sample backwards in time.
That gives us the following figure.
- - ------------------------|
codebook memory |
- - ------------------------|
|-5-|---16---|-5-|
pi pp po
| | Codebook vector
|---16---|-5-|-----19-----| <- corresponding to
i ii iii sample delay 21
Figure 3.12. Generation of the second augmented codebook
The figure 3.12 shows the codebook memory with pointers pi, pp and
po where pi points to sample 26, pp to sample 21 and po to sample 5.
Below the codebook memory, the augmented codebook vector
corresponding to sample delay 21 is drawn. Segment i does now
consist of 16 samples from pp and forward. Segment ii consists of 5
interpolated samples from pi and forward and po and forward and the
interpolation weights are the same throughout the procedure. Segment
iii consists of 19 samples from pp and forward. The same procedure
of moving the two pointers is continued until the last augmented
vector corresponding to sample delay 39 has been created. This gives
a total of 20 new codebook vectors to each of the two sections. Thus
the total number of codebook vectors for each of the two sections,
when including the augmented codebook becomes lMem-SUBL+1+SUBL/2.
This is provided that augmentation is evoked, i.e., that
lTarget=SUBL.
3.6.4 Codebook Search
The codebook search uses the codebooks described in the sections
above to find the best match of the perceptually weighted target,
see section 3.6.2. The search method is a multi-stage gain-shape
matching performed as follows. At each stage the best shape vector
is identified, then the gain is calculated and quantized, and
finally the target is updated in preparation for the next codebook
search stage. The number of stages is CB_NSTAGES=3.
If the target is the 23/22 sample vector the codebooks are indexed
in the order: base codebook followed by the expanded codebook. If
the target is 40 samples the order is: base codebook, augmented base
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codebook, expanded codebook and finally augmented expanded codebook.
The size of each codebook section and its corresponding augmented
section is given by table 3.1 in section 3.6.3.
For example when coding the second 40 sample sub-block indices 0-107
correspond to the base codebook, 108-127 correspond to the augmented
base codebook, 128-235 correspond to the expanded codebook and
finally indices 236-255 correspond to the augmented expanded
codebook. The indices are divided in the same fashion for all stages
in the example. Only in the case of coding the first 40 sample sub-
block is there a difference between stages (see Table 3.1).
3.6.4.1 Codebook Search at Each Stage
The codebooks are searched to find the best match to the target at
each stage. When the best match is found the target is updated and
the next-stage search is started. The three chosen codebook vectors
and their corresponding gains constitute the encoded sub-block. The
best match is decided by the following three criteria:
1. Compute the measure
(target*cbvec)^2 / ||cbvec||^2
for all codebook vectors, cbvec, and choose the codebook vector
maximizing the measure. The expression (target*cbvec) is the dot
product between the target vector to be coded and the codebook
vector for which we compute the measure. The norm, ||x||, is defined
as the square root of (x*x).
2. The absolute value of the gain, corresponding to the chosen
codebook vector, cbvec, must be smaller than a fixed limit,
CB_MAXGAIN=1.3:
|gain| < CB_MAXGAIN
where the gain is computed in the following way:
gain = (target*cbvec) / ||cbvec||^2
3. For the first stage the dot product of the chosen codebook vector
and target must be positive:
target*cbvec > 0
In practice the above criteria are used in a sequential search
through all codebook vectors. The best match is found by registering
a new max measure and index whenever the previously registered max
measure is surpassed and all other criteria are fulfilled. If none
of the codebook vectors fulfill (2) and (3), the first codebook
vector is selected.
3.6.4.2 Gain Quantization at Each Stage
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The gain follows as a result of the computation:
gain = (target*cbvec) / ||cbvec||^2
for the optimal codebook vector that was found by the procedure from
section 3.6.4.1.
The three stages quantize the gain using 5, 4 and 3 bits
respectively. In the first stage, the gain is limited to positive
values. This gain is quantized by finding the nearest value in the
quantization table gain_sq5Tbl.
gain_sq5Tbl[32]={0.037476, 0.075012, 0.112488, 0.150024, 0.187500,
0.224976, 0.262512, 0.299988, 0.337524, 0.375000,
0.412476, 0.450012, 0.487488, 0.525024, 0.562500,
0.599976, 0.637512, 0.674988, 0.712524, 0.750000,
0.787476, 0.825012, 0.862488, 0.900024, 0.937500,
0.974976, 1.012512, 1.049988, 1.087524, 1.125000,
1.162476, 1.200012}
The gains of the subsequent two stages can be either positive or
negative. The gains are quantized using a quantization table times a
scale factor. The second stage uses the table gain_sq4Tbl and the
third stage uses gain_sq3Tbl. The scale factor equates 0.1 or the
absolute value of the quantized gain representation value obtained
in the previous stage, whichever is the larger. Again, the resulting
gain index is the index to the nearest value of the quantization
table times the scale factor.
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gainQ = scaleFact * gain_sqXTbl[index]
gain_sq4Tbl[16]={-1.049988, -0.900024, -0.750000, -0.599976,
-0.450012, -0.299988, -0.150024, 0.000000, 0.150024,
0.299988, 0.450012, 0.599976, 0.750000, 0.900024,
1.049988, 1.200012}
gain_sq3Tbl[8]={-1.000000, -0.659973, -0.330017,0.000000,
0.250000, 0.500000, 0.750000, 1.00000}
3.6.4.3 Preparation of Target for Next Stage
Before performing the search for the next stage the perceptually
weighted target vector is updated by subtracting from it the
selected codebook vector (from the perceptually weighted codebook)
times the corresponding quantized gain.
target[i] = target[i] - gainQ * selected_vec[i];
A reference implementation of the codebook encoding is found in
Appendix A.34.
3.7 Gain Correction Encoding
The start state is quantized in a relatively model independent
manner using 3 bits per sample. In contrast to this, the remaining
parts of the block is encoded using an adaptive codebook. This
codebook will produce high matching accuracy whenever there is a
high correlation between the target and the best codebook vector.
For unvoiced speech segments and background noises, this is not
necessarily so, which, due to the nature of the squared error
criterion, results in a coded signal with less power than the target
signal. As the coded start state has good power matching to the
target, the result is a power fluctuation within the encoded frame.
Perceptually, the main problem with this is that the time envelope
of the signal energy becomes unsteady. To overcome this problem, the
gains for the codebooks are re-scaled after the codebook encoding by
searching for a new gain factor for the first stage codebook that
provides better power matching.
First the energy for the target signal, tene, is computed along with
the energy for the coded signal, cene, given by the addition of the
3 gain scaled codebook vectors. Since the gains of the 2nd and 3rd
stage scale with the gain of the first stage, by changing the first
stage gain from gain[0] to gain_sq5Tbl[i], the energy of the coded
signal changes from cene to
cene*(gain_sq5Tbl[i]*gain_sq5Tbl[i])/(gain[0]*gain[0])
where gain[0] is the gain for the first stage found in the original
codebook search. A refined search is performed by testing the gain
indices i=0 to 31, and as long as the new codebook energy as given
above is less than tene, the gain index for stage 1 is increased. A
restriction is applied so that the new gain value for stage 1 cannot
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be more than 2 times higher than the original value found in the
codebook search. Note that by using this method the shape of the
encoded vector is not changed, only the gain or amplitude.
3.8 Bitstream Definition
The total number of bits used to describe one frame of 20 ms speech
is 304, which fits in 38 bytes and results in a bit rate of 15.20
kbit/s. For the case with a frame length of 30 ms speech the total
number of bits used is 400, which fits in 50 bytes and results in a
bit rate of 13.33 kbit/s. In the bitstream definition the bits are
distributed into three classes according to their bit error or loss
sensitivity. The most sensitive bits (class 1) are placed first in
the bitstream for each frame. The less sensitive bits (class 2) are
placed after the class 1 bits. The least sensitive bits (class 3)
are placed at the end of the bitstream for each frame.
Looking at the 20/30 ms frame length cases for each class: The class
1 bits occupy a total of 6/8 bytes (48/64 bits), the class 2 bits
occupy 8/12 bytes (64/96 bits), and the class 3 bits occupy 24/30
bytes (191/239 bits). This distribution of the bits enable the use
of uneven level protection (ULP) as is exploited in the payload
format definition for iLBC [1]. The detailed bit allocation is shown
in the table below. When a quantization index is distributed between
more classes the more significant bits belong to the lowest class.
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Bitstream structure:
------------------------------------------------------------------+
Parameter | Bits Class <1,2,3> |
| 20 ms frame | 30 ms frame |
----------------------------------+---------------+---------------+
Split 1 | 6 <6,0,0> | 6 <6,0,0> |
LSF 1 Split 2 | 7 <7,0,0> | 7 <7,0,0> |
LSF Split 3 | 7 <7,0,0> | 7 <7,0,0> |
------------------+---------------+---------------+
Split 1 | NA (Not Appl.)| 6 <6,0,0> |
LSF 2 Split 2 | NA | 7 <7,0,0> |
Split 3 | NA | 7 <7,0,0> |
------------------+---------------+---------------+
Sum | 20 <20,0,0> | 40 <40,0,0> |
----------------------------------+---------------+---------------+
Block Class. | 2 <2,0,0> | 3 <3,0,0> |
----------------------------------+---------------+---------------+
Position 22 sample segment | 1 <1,0,0> | 1 <1,0,0> |
----------------------------------+---------------+---------------+
Scale Factor State Coder | 6 <6,0,0> | 6 <6,0,0> |
----------------------------------+---------------+---------------+
Sample 0 | 3 <0,1,2> | 3 <0,1,2> |
Quantized Sample 1 | 3 <0,1,2> | 3 <0,1,2> |
Residual : | : : | : : |
State : | : : | : : |
Samples : | : : | : : |
Sample 56 | 3 <0,1,2> | 3 <0,1,2> |
Sample 57 | NA | 3 <0,1,2> |
------------------+---------------+---------------+
Sum | 171 <0,57,114>| 174 <0,58,116>|
----------------------------------+---------------+---------------+
Stage 1 | 7 <6,0,1> | 7 <4,2,1> |
CB for 22/23 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
sample block Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
------------------+---------------+---------------+
Sum | 21 <6,0,15> | 21 <4,2,15> |
----------------------------------+---------------+---------------+
Stage 1 | 5 <2,0,3> | 5 <1,1,3> |
Gain for 22/23 Stage 2 | 4 <1,1,2> | 4 <1,1,2> |
sample block Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
------------------+---------------+---------------+
Sum | 12 <3,1,8> | 12 <2,2,8> |
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----------------------------------+---------------+---------------+
Stage 1 | 8 <7,0,1> | 8 <6,1,1> |
sub-block 1 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
------------------+---------------+---------------+
Stage 1 | 8 <0,0,8> | 8 <0,7,1> |
sub-block 2 Stage 2 | 8 <0,0,8> | 8 <0,0,8> |
Indices Stage 3 | 8 <0,0,8> | 8 <0,0,8> |
for CB ------------------+---------------+---------------+
sub-blocks Stage 1 | NA | 8 <0,7,1> |
sub-block 3 Stage 2 | NA | 8 <0,0,8> |
Stage 3 | NA | 8 <0,0,8> |
------------------+---------------+---------------+
Stage 1 | NA | 8 <0,7,1> |
sub-block 4 Stage 2 | NA | 8 <0,0,8> |
Stage 3 | NA | 8 <0,0,8> |
------------------+---------------+---------------+
Sum | 46 <7,0,39> | 94 <6,22,66> |
----------------------------------+---------------+---------------+
Stage 1 | 5 <1,2,2> | 5 <1,2,2> |
sub-block 1 Stage 2 | 4 <1,1,2> | 4 <1,2,1> |
Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
------------------+---------------+---------------+
Stage 1 | 5 <1,1,3> | 5 <0,2,3> |
sub-block 2 Stage 2 | 4 <0,2,2> | 4 <0,2,2> |
Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
Gains for ------------------+---------------+---------------+
sub-blocks Stage 1 | NA | 5 <0,1,4> |
sub-block 3 Stage 2 | NA | 4 <0,1,3> |
Stage 3 | NA | 3 <0,0,3> |
------------------+---------------+---------------+
Stage 1 | NA | 5 <0,1,4> |
sub-block 4 Stage 2 | NA | 4 <0,1,3> |
Stage 3 | NA | 3 <0,0,3> |
------------------+---------------+---------------+
Sum | 24 <3,6,15> | 48 <2,12,34> |
----------------------------------+---------------+---------------+
Empty frame indicator | 1 <0,0,1> | 1 <0,0,1> |
-------------------------------------------------------------------
SUM 304 <48,64,192> 400 <64,96,240>
Table 3.2. The bitstream definition for iLBC for both the 20 ms
frame size mode and the 30 ms frame size mode.
When packetized into the payload the bits MUST be sorted as: All the
class 1 bits in the order (from top to bottom) as they were
specified in the table, all the class 2 bits (from top to bottom)
and finally all the class 3 bits in the same sequential order.
The last bit, the empty frame indicator, SHOULD be set to zero by
the encoder. If this bit is set to one the decoder SHOULD treat the
data as a lost frame. For example this bit can be set to 1 to
indicate lost frame for file storage format as in [1].
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4. DECODER PRINCIPLES
This section describes the principles of each component of the
decoder algorithm.
+-------------+ +--------+ +---------------+
payload -> | 1. Get para | -> | 2. LPC | -> | 3. Sc Dequant | ->
+-------------+ +--------+ +---------------+
+-------------+ +------------------+
-> | 4 Mem setup | -> | 5. Construct res |------->
| +-------------+ +------------------- |
---------<-----------<-----------<------------
Sub-frame 0...2/4 (20 ms/30 ms)
+----------------+ +----------+
-> | 6. Enhance res | -> | 7. Synth | ------------>
+----------------+ +----------+
+-----------------+
-> | 8. Post Process | ----------------> decoded speech
+-----------------+
Figure 4.1. Flow chart of the iLBC decoder. If a frame was lost
steps 1 to 5 SHOULD be replaced by a PLC algorithm.
1. Extract the parameters from the bitstream
2. Decode the LPC and interpolate (section 4.1)
3. Construct the 57/58 sample start state (section 4.2)
4. Set up the memory using data from the decoded residual. This
memory is used for codebook construction. For blocks preceding the
start state, both the decoded residual and the target are time
reversed. Sub-frames are decoded in the same order as they were
encoded
5. Construct the residuals of this sub-frame (gain[0]*cbvec[0] +
gain[1]*cbvec[1] + gain[2]*cbvec[2]). Repeat 4 and 5 until the
residual of all sub-blocks have been constructed
6. Enhance the residual with the post filter (section 4.6)
7. Synthesis of the residual (section 4.7)
8. Post process with HP filter if desired (section 4.8)
4.1 LPC Filter Reconstruction
The decoding of the LP filter parameters is very straightforward.
For a set of three/six indices the corresponding LSF vector(s) are
found by simple table look up. For each of the LSF vectors the three
split vectors are concatenated to obtain qlsf1 and qlsf2,
respectively (in the 20 ms mode only one LSF vector, qlsf, is
constructed). The next step is the stability check described in
Section 3.2.5 followed by the interpolation scheme described in
Section 3.2.6 (3.2.7 for 20 ms frames). The only difference is that
only the quantized LSFs are known at the decoder and hence the
unquantized LSFs are not processed.
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A reference implementation of the LPC filter reconstruction is given
in Appendix A.36.
4.2 Start State Reconstruction
The scalar encoded STATE_SHORT_LEN=58 (STATE_SHORT_LEN=57 in the 20
ms mode) state samples are reconstructed by 1) forming a set of
samples (by table look-up) from the index stream idxVec[n] 2)
multiplying the set with 1/scal=(10^qmax)/4.5 3) time reversing the
57/58 samples 4) filtering the time reversed block with the
dispersion (all-pass) filter used in the encoder (as described in
section 3.5.2). This compensates for the phase distortion of the
earlier filter operation. 5) Reversing the 57/58 samples from the
previous step
in(0..(STATE_SHORT_LEN-1)) = time reversed samples from table
look-up,
idxVecDec((STATE_SHORT_LEN-1)..0)
in(STATE_SHORT_LEN..(2*STATE_SHORT_LEN-1)) = 0
Pk(z) = A~rk(z)/A~k(z), where
___
\
A~rk(z)= z^(-LPC_FILTERORDER) + > a~ki*z^(i-(LPC_FILTERORDER-1))
/__
i=0...(LPC_FILTERORDER-1)
and A~k(z) is taken from the block where the start state begins
in -> Pk(z) -> filtered
out(k) = filtered(STATE_SHORT_LEN-1-k) +
filtered(2*STATE_SHORT_LEN-1-k),
k=0..(STATE_SHORT_LEN-1)
The remaining 23/22 samples in the state are reconstructed by the
same adaptive codebook technique as described in section 4.3. The
location bit determines whether these are the first or the last
23/22 samples of the 80 sample state vector. If the remaining 23/22
samples are the first samples of the state vector, then the scalar
encoded STATE_SHORT_LEN state samples are time-reversed before
initialization of the adaptive codebook memory vector.
A reference implementation of the start state reconstruction is
given in Appendix A.44.
4.3 Excitation Decoding Loop
The decoding of the LPC excitation vector proceeds in the same order
in which the residual was encoded at the encoder. That is, after the
decoding of the entire 80 sample state vector, the forward sub-
blocks (corresponding to samples occurring after the state vector
samples) are decoded, and then the backward sub-blocks
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(corresponding to samples occurring before the state vector) are
decoded, resulting in a fully decoded block of excitation signal
samples.
In particular, each sub-block is decoded using the multistage
adaptive codebook decoding module which is described in section 4.4.
This module relies upon an adaptive codebook memory that is
constructed before each run of the adaptive codebook decoding. The
construction of the adaptive codebook memory in the decoder is
identical to the method outlined in section 3.6.3, except that it is
done on the codebook memory without perceptual weighting.
For the initial forward sub-block, the last STATE_LEN=80 samples of
the length CB_LMEM=147 adaptive codebook memory are filled with the
samples of the state vector. For subsequent forward sub-blocks, the
first SUBL=40 samples of the adaptive codebook memory are discarded,
the remaining samples are shifted by SUBL samples towards the
beginning of the vector, while the newly decoded SUBL=40 samples are
placed at the end of the adaptive codebook memory. For backward sub-
blocks, the construction is similar except that every vector of
samples involved is first time-reversed.
A reference implementation of the excitation decoding loop is found
in Appendix A.5.
4.4 Multistage Adaptive Codebook Decoding
The Multistage Adaptive Codebook Decoding module is used at both the
sender (encoder) and the receiver (decoder) ends to produce a
synthetic signal in the residual domain that is eventually used to
produce synthetic speech. The module takes the index values used to
construct vectors that are scaled and summed together to produce a
synthetic signal that is the output of the module.
4.4.1 Construction of the Decoded Excitation Signal
The unpacked index values provided at the input to the module are
references to extended codebooks, which are constructed as described
in Section 3.6.3 with the only difference that it is based on the
codebook memory without the perceptual weighting. The unpacked 3
indices are used to look up 3 codebook vectors. The unpacked 3 gain
indices are used to decode the corresponding 3 gains. In this
decoding the successive rescaling as described in Section 3.6.4.2 is
applied.
A reference implementation of the adaptive codebook decoding is
listed in Appendix A.32.
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4.5 Packet Loss Concealment
If packet loss occurs, the decoder receives a signal saying that
information regarding a block is lost. For such blocks it is
RECOMMENDED to use a Packet Loss Concealment (PLC) unit to create a
decoded signal which masks the effect of that packet loss. In the
following we will describe an example of a PLC unit that can be used
with the iLBC codec. As the PLC unit is used only at the decoder,
the PLC unit does not affect interoperability between
implementations. Other PLC implementations MAY therefore be used.
The example PLC described operates on the LP filters and the
excitation signals and is based on the following principles:
4.5.1 Block Received Correctly and Previous Block also Received
If the block is received correctly, the PLC only records state
information of the current block that can be used in case the next
block is lost. The LP filter coefficients for each sub-block and the
entire decoded excitation signal are all saved in the decoder state
structure. All this information will be needed if the following
block is lost.
4.5.2 Block Not Received
If the block is not received, the block substitution is based on
doing a pitch synchronous repetition of the excitation signal which
is filtered by the last LP filter of the previous block. The
previous block's information is stored in the decoder state
structure.
A correlation analysis is performed on the previous block's
excitation signal in order to detect the amount of pitch periodicity
and a pitch value. The correlation measure is also used to decide on
the voicing level (the degree to which the previous block's
excitation was a voiced or roughly periodic signal). The excitation
in the previous block is used to create an excitation for the block
to be substituted such that the pitch of the previous block is
maintained. Therefore, the new excitation is constructed in a pitch
synchronous manner. In order to avoid a buzzy sounding substituted
block, a random excitation is mixed with the new pitch periodic
excitation and the relative use of the two components is computed
from the correlation measure (voicing level).
For the block to be substituted, the newly constructed excitation
signal is then passed through the LP filter to produce the speech
that will be substituted for the lost block.
For several consecutive lost blocks, the packet loss concealment
continues in a similar manner. The correlation measure of the last
received block is still used along with the same pitch value. The LP
filters of the last received block are also used again. The energy
of the substituted excitation for consecutive lost blocks is
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decreased, leading to a dampened excitation, and therefore dampened
speech.
4.5.3 Block Received Correctly When Previous Block Not Received
For the case in which a block is received correctly when the
previous block was not received, the correctly received block's
directly decoded speech (based solely on the received block) is not
used as the actual output. The reason for this is that the directly
decoded speech does not necessarily smoothly merge into the
synthetic speech generated for the previous lost block. If the two
signals are not smoothly merged, an audible discontinuity is
accidentally produced. Therefore, a correlation analysis between the
two blocks of excitation signal (the excitation of the previous
concealed block and the excitation of the current received block) is
performed to find the best phase match. Then a simple overlap-add
procedure is performed to smoothly merge the previous excitation
into the current block's excitation.
The exact implementation of the packet loss concealment does not
influence interoperability of the codec.
A reference implementation of the packet loss concealment is
suggested in Appendix A.14. Exact compliance with this suggested
algorithm is not needed for a reference implementation to be fully
compatible with the overall codec specification.
4.6 Enhancement
The decoder contains an enhancement unit that operates on the
reconstructed excitation signal. The enhancement unit increases the
perceptual quality of the reconstructed signal by reducing the
speech-correlated noise in the voiced speech segments. Compared to
traditional postfilters, the enhancer has the advantage that it can
only modify the excitation signal slightly. This means that there is
no risk of over enhancement. The enhancer works very similar for
both the 20 ms frame size mode and for the 30 ms frame size mode.
For the mode with 20 ms frame size, the enhancer uses a memory of
six 80 sample excitation blocks prior in time plus the two new 80
sample excitation blocks. For each block of 160 new unenhanced
excitation samples, 160 enhanced excitation samples are produced.
The enhanced excitation is 40 sample delayed compared to the
unenhanced excitation since the enhancer algorithm uses lookahead.
For the mode with 30 ms frame size, the enhancer uses a memory of
five 80 sample excitation blocks prior in time plus the three new 80
sample excitation blocks. For each block of 240 new unenhanced
excitation samples, 240 enhanced excitation samples are produced.
The enhanced excitation is 80 sample delayed compared to the
unenhanced excitation since the enhancer algorithm uses lookahead.
OUTLINE of Enhancer
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The speech enhancement unit operates on sub-blocks of 80 samples,
which means that there are two/three 80 sample sub-blocks per frame.
Each of these two/three sub-blocks is enhanced separately, but in an
analogous manner.
unenhanced residual
|
| +---------------+ +--------------+
+-> | 1. Pitch Est | -> | 2. Find PSSQ | -------->
+---------------+ | +--------------+
+-----<-------<------<--+
+------------+ enh block 0..1/2 |
-> | 3. Smooth | |
+------------+ |
\ |
/\ |
/ \ Already |
/ 4. \----------->----------->-----------+ |
\Crit/ Fulfilled | |
\? / v |
\/ | |
\ +-----------------+ +---------+ | |
Not +->| 5. Use Constr. | -> | 6. Mix | ----->
Fulfilled +-----------------+ +---------+
---------------> enhanced residual
Figure 4.2. Flow chart of the enhancer
1. Pitch estimation of each of the two/three new 80 sample blocks
2. Find the pitch-period-synchronous sequence n (for block k) by a
search around the estimated pitch value. Do this for
n=1,2,3,-1,-2,-3
3. Calculate the smoothed residual generated by the 6 pitch-period-
synchronous sequences from prior step
4. Check if the smoothed residual satisfies the criterion (section
4.6.4)
5. Use constraint to calculate mixing factor (section 4.6.5)
6. Mix smoothed signal with unenhanced residual (pssq(n) n=0)
The main idea of the enhancer is to find three 80 sample blocks
before and three 80 sample blocks after the analyzed unenhanced sub-
block and use these to improve the quality of the excitation in that
sub-block. The six blocks are chosen so that they have the highest
possible correlation with the unenhanced sub-block that is being
enhanced. In other words the 6 blocks are pitch-period-synchronous
sequences to the unenhanced sub-block.
A linear combination of the six pitch-period-synchronous sequences
is calculated that approximates the sub-block. If the squared error
between the approximation and the unenhanced sub-block is small
enough, the enhanced residual is set equal to this approximation.
For the cases when the squared error criterion is not fulfilled, a
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linear combination of the approximation and the unenhanced residual
forms the enhanced residual.
4.6.1 Estimating the pitch
Pitch estimates are needed to determine the locations of the pitch-
period-synchronous sequences in a complexity efficient way. For each
of the new two/three sub-blocks a pitch estimate is calculated by
finding the maximum correlation in the range from lag 20 to lag 120.
These pitch estimates are used to narrow down the search for the
best possible pitch-period-synchronous sequences.
4.6.2 Determination of the Pitch-Synchronous Sequences
Upon receiving the pitch estimates from the prior step, the enhancer
analyzes and enhances one 80 sample sub-block at a time. The pitch-
period-synchronous-sequences pssq(n) can be viewed as vectors of
length 80 samples each shifted n*lag samples from the current sub-
block. The six pitch-period-synchronous-sequences, pssq(-3) to
pssq(-1) and pssq(1) to pssq(3), are found one at a time by the
steps below:
1) Calculate the estimate of the position of the pssq(n). For
pssq(n) in front of pssq(0) (n > 0), the location of the pssq(n)
is estimated by moving one pitch estimate forward in time from
the exact location of pssq(n-1). Similarly for pssq(n) behind
pssq(0) (n < 0) is estimated by moving one pitch estimate
backward in time from the exact location of pssq(n+1). If the
estimated pssq(n) vector location is totally within the enhancer
memory (figure 4.3) step 2,3, and 4 are performed, otherwise the
pssq(n) is set to zeros.
2) Compute the correlation between the unenhanced excitation and
vectors around the estimated location interval of pssq(n). The
correlation is calculated in the interval estimated location +/-
2 samples. This results in 5 correlation values.
3) The 5 correlation values are upsampled by a factor 4, using sinc
upsampling filters (four MA filters with coefficients upsFilter1
.. upsFilter4). Within these the maximum value is found, which
specifies the best pitch-period with a resolution of a quarter of
a sample.
upsFilter1[7]={0.000000 0.000000 0.000000 1.000000
0.000000 0.000000 0.000000}
upsFilter2[7]={0.015625 -0.076904 0.288330 0.862061
-0.106445 0.018799 -0.015625}
upsFilter3[7]={0.023682 -0.124268 0.601563 0.601563
-0.124268 0.023682 -0.023682}
upsFilter4[7]={0.018799 -0.106445 0.862061 0.288330
-0.076904 0.015625 -0.018799}
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4) Generate the pssq(n) vector by upsampling of the excitation
memory and extracting the sequence that corresponds to the lag
delay that was calculated in prior step.
With the steps above all the pssq(n) can be found in an iterative
manner, first moving backward in time from pssq(0) and then forward
in time from pssq(0).
0 159 319 479 639
+---------------------------------------------------------------+
| -5 | -4 | -3 | -2 | -1 | 0 | 1 | 2 |
+---------------------------------------------------------------+
|pssq 0 |
|pssq -1| |pssq 1 |
|pssq -2| |pssq 2 |
|pssq -3| |pssq 3 |
Figure 4.3. Enhancement for 20 ms frame size
Figure 4.3. depicts pitch-period-synchronous sequences in the
enhancement of the first 80 sample block in the 20 ms frame size
mode. The unenhanced signal input is stored in the two last sub-
blocks (1-2), and the six other sub-blocks contain unenhanced
residual prior-in-time. We perform the enhancement algorithm on two
blocks of 80 samples, where the first of the two blocks consist of
the last 40 samples of sub-block 0 and the first 40 samples of sub-
block 1. The second 80 sample block consists of the last 40 samples
of sub-block 1 and the first 40 samples of sub-block 2.
0 159 319 479 639
+---------------------------------------------------------------+
| -4 | -3 | -2 | -1 | 0 | 1 | 2 | 3 |
+---------------------------------------------------------------+
|pssq 0 |
|pssq -1| |pssq 1 |
|pssq -2| |pssq 2 |
|pssq -3| |pssq 3 |
Figure 4.4. Enhancement for 30 ms frame size
Figure 4.4. depicts pitch-period-synchronous sequences in the
enhancement of the first 80 sample block in the 30 ms frame size
mode. The unenhanced signal input is stored in the three last sub-
blocks (1-3). The five other sub-blocks contain unenhanced residual
prior-in-time. The enhancement algorithm is performed on the three
80 sample sub-blocks 0, 1 and 2.
4.6.3 Calculation of the smoothed excitation
A linear combination of the six pssq(n) (n!=0) form a smoothed
approximation, z, of pssq(0). Most of the weight is put on the
sequences that are close to pssq(0) since these are most likely to
be most similar to pssq(0). The smoothed vector is also rescaled, so
that the energy of z is the same as the energy of pssq(0).
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___
\
y = > pssq(i) * pssq_weight(i)
/__
i=-3,-2,-1,1,2,3
pssq_weight(i) = 0.5*(1-cos(2*pi*(i+4)/(2*3+2)))
z = C * y, where C = ||pssq(0)||/||y||
4.6.4 Enhancer criterion
The criterion of the enhancer is that the enhanced excitation is not
allowed to differ much from the unenhanced excitation. This
criterion is checked for each 80 sample sub-block.
e < (b * ||pssq(0)||^2), where b=0.05 and (Constraint 1)
e = (pssq(0)-z)*(pssq(0)-z), and "*" means the dot product
4.6.5 Enhancing the excitation
From the criterion in the previous section it is clear that the
excitation is not allowed to change much. The purpose of this
constraint is to prevent the creation of an enhanced signal that is
significantly different from the original signal. This also means
that the constraint limits the numerical size of the errors that the
enhancement procedure can make. That is especially important in
unvoiced segments and background noise segments where increased
periodicity could lead to lower perceived quality.
When the constraint in the prior section is not met, the enhanced
residual is instead calculated through a constrained optimization
using the Lagrange multiplier technique. The new constraint is that:
e = (b * ||pssq(0)||^2) (Constraint 2)
We distinguish two solution regions for the optimization: 1) the
region where the first constraint is fulfilled and 2) the region
where the first constraint is not fulfilled so the second constraint
must be used.
In the first case, where the second constraint is not needed, the
optimized re-estimated vector is simply z, the energy scaled version
of y.
In the second case, where the second constraint is activated and
becomes an equality constraint, we have that
z= A*y + B*pssq(0)
where
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A = sqrt((b-b^2/4)*(w00*w00)/ (w11*w00 + w10*w10)) and
w11 = pssq(0)*pssq(0)
w00 = y*y
w10 = y*pssq(0) (* symbolizes the dot product)
and
B = 1 - b/2 - A * w10/w00
Appendix A.16 contains a listing of a reference implementation for
the enhancement method.
4.7 Synthesis Filtering
Upon decoding or PLC of the LP excitation block, the decoded speech
block is obtained by running the decoded LP synthesis filter,
1/A~k(z), over the block. The synthesis filters have to be shifted
to compensate for the delay in the enhancer. For 20 ms frame size
mode they SHOULD be shifted one 40 sample sub-block and for 30 ms
frame size mode they SHOULD be shifted two 40 sample sub-blocks. The
LP coefficients SHOULD be changed at the first sample of every sub-
block while keeping the filter state. For PLC blocks, one solution
is to apply the last LP coefficients of the last decoded speech
block for all sub-blocks.
The reference implementation for the synthesis filtering can be
found in Appendix A.48.
4.8 Post Filtering
If desired the decoded block can be filtered by a high-pass filter.
This removes the low frequencies of the decoded signal. A reference
implementation of this, with cut off at 65 Hz, is shown in Appendix
A.30.
5. IANA CONSIDERATIONS
This algorithm for the coding of speech signals does not have any
IANA considerations that need to be addressed.
6. SECURITY CONSIDERATIONS
This algorithm for the coding of speech signals is not subject of
any known security consideration; however, its RTP payload format
[1] is subject of several considerations which are addressed there.
7. EVALUATION OF THE ILBC IMPLEMENTATIONS
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It is possible and suggested to evaluate certain iLBC implementation
by utilizing methodology and tools available at
http://www.ilbcfreeware.org/evaluation.html
8. REFERENCES
8.1 Normative
[1] A. Duric and S. V. Andersen, "RTP Payload Format for iLBC
Speech", IETF Draft, October 2003.
[2] S. Bradner, "Key words for use in RFCs to Indicate requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] PacketCable(TM) Audio/Video Codecs Specification, Cable
Television Laboratories, Inc.
8.2 Informative
[4] ITU-T Recommendation G.711, available online from the ITU
bookstore at http://www.itu.int.
9. ACKNOWLEDGEMENTS
The authors are deeply indebted to them all and thank them
sincerely:
Henry Sinnreich, Patrik Faltstrom and Alan Johnston
for great support of the iLBC initiative and for valuable feedback
and comments.
Peter Vary, Frank Mertz and Christoph Erdmann (RWTH Aachen);
Vladimir Cuperman (Niftybox LLC); Thomas Eriksson (Chalmers
Univ of Tech) and Gernot Kubin (TU Graz)
for thorough review of the iLBC draft and their valuable feedback
and remarks.
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10. AUTHOR'S ADDRESSES
Soren Vang Andersen
Department of Communication Technology
Aalborg University
Fredrik Bajers Vej 7A
9200 Aalborg
Denmark
Phone: ++45 9 6358627
Email: sva@kom.auc.dk
Henrik Astrom
Global IP Sound AB
Rosenlundsgatan 54
Stockholm, S-11863
Sweden
Phone: +46 8 54553040
Email: henrik.astrom@globalipsound.com
Alan Duric
Global IP Sound AB
Rosenlundsgatan 54
Stockholm, S-11863
Sweden
Phone: +46 8 54553040
Email: alan.duric@globalipsound.com
Fredrik Galschiodt
Global IP Sound AB
Rosenlundsgatan 54
Stockholm, S-11863
Sweden
Phone: +46 8 54553040
Email: fredrik.galschiodt@globalipsound.com
Roar Hagen
Global IP Sound AB
Rosenlundsgatan 54
Stockholm, S-11863
Sweden
Phone: +46 8 54553040
Email: roar.hagen@globalipsound.com
W. Bastiaan Kleijn
Global IP Sound AB
Rosenlundsgatan 54
Stockholm, S-11863
Sweden
Phone: +46 8 54553040
Email: bastiaan.kleijn@globalipsound.com
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Jan Linden
Global IP Sound Inc.
900 Kearny Street, suite 500
San Francisco, CA-94133
USA
Phone: +1 415 397 2555
Email: jan.linden@globalipsound.com
Manohar N. Murthi
Department of Electrical and Computer Engineering
University of Miami
McArthur Engineering Building 406
1251 Memorial Dr
Coral Gables, FL 33146-0640
USA
Phone: +1 (305) 284-3342
Email: mmurthi@miami.edu
Jan Skoglund
Global IP Sound Inc.
900 Kearny Street, suite 500
San Francisco, CA-94133
USA
Phone: +1 415 397 2555
Email: jan.skoglund@globalipsound.com
Julian Spittka
Global IP Sound Inc.
900 Kearny Street, suite 500
San Francisco, CA-94133
USA
Phone: +1 415 397 2555
Email: julian.spittka@globalipsound.com
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APPENDIX A REFERENCE IMPLEMENTATION
This appendix contains the complete c-code for a reference
implementation of encoder and decoder for the specified codec.
The c-code consists of the following files with highest level
functions:
iLBC_test.c: main function for evaluation purpose
iLBC_encode.h: encoder header
iLBC_encode.c: encoder function
iLBC_decode.h: decoder header
iLBC_decode.c: decoder function
the following files containing global defines and constants:
iLBC_define.h: global defines
constants.h: global constants header
constants.c: global constants memory allocations
and the following files containing subroutines:
anaFilter.h: lpc analysis filter header
anaFilter.c: lpc analysis filter function
createCB.h: codebook construction header
createCB.c: codebook construction function
doCPLC.h: packet loss concealment header
doCPLC.c: packet loss concealment function
enhancer.h: signal enhancement header
enhancer.c: signal enhancement function
filter.h: general filter header
filter.c: general filter functions
FrameClassify.h: start state classification header
FrameClassify.c: start state classification function
gainquant.h: gain quantization header
gainquant.c: gain quantization function
getCBvec.h: codebook vector construction header
getCBvec.c: codebook vector construction function
helpfun.h: general purpose header
helpfun.c: general purpose functions
hpInput.h: input high pass filter header
hpInput.c: input high pass filter function
hpOutput.h: output high pass filter header
hpOutput.c: output high pass filter function
iCBConstruct.h: excitation decoding header
iCBConstruct.c: excitation decoding function
iCBSearch.h: excitation encoding header
iCBSearch.c: excitation encoding function
LPCdecode.h: lpc decoding header
LPCdecode.c: lpc decoding function
LPCencode.h: lpc encoding header
LPCencode.c: lpc encoding function
lsf.h: line spectral frequencies header
lsf.c: line spectral frequencies functions
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packing.h: bitstream packetization header
packing.c: bitstream packetization functions
StateConstructW.h: state decoding header
StateConstructW.c: state decoding functions
StateSearchW.h: state encoding header
StateSearchW.c: state encoding function
syntFilter.h: lpc synthesis filter header
syntFilter.c: lpc synthesis filter function
The implementation is portable and should work on many different
platforms. However, it is not difficult to optimize the
implementation on particular platforms, an exercise left to the
reader.
A.1 iLBC_test.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_test.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include "iLBC_define.h"
#include "iLBC_encode.h"
#include "iLBC_decode.h"
/* Runtime statistics */
#include <time.h>
#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS/2)
/*----------------------------------------------------------------*
* Encoder interface function
*---------------------------------------------------------------*/
short encode( /* (o) Number of bytes encoded */
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i/o) Encoder instance */
short *encoded_data, /* (o) The encoded bytes */
short *data /* (i) The signal block to encode*/
){
float block[BLOCKL_MAX];
int k;
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/* convert signal to float */
for (k=0; k<iLBCenc_inst->blockl; k++)
block[k] = (float)data[k];
/* do the actual encoding */
iLBC_encode((unsigned char *)encoded_data, block, iLBCenc_inst);
return (iLBCenc_inst->no_of_bytes);
}
/*----------------------------------------------------------------*
* Decoder interface function
*---------------------------------------------------------------*/
short decode( /* (o) Number of decoded samples */
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
short *decoded_data, /* (o) Decoded signal block*/
short *encoded_data, /* (i) Encoded bytes */
short mode /* (i) 0=PL, 1=Normal */
){
int k;
float decblock[BLOCKL_MAX], dtmp;
/* check if mode is valid */
if (mode<0 || mode>1) {
printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
/* do actual decoding of block */
iLBC_decode(decblock, (unsigned char *)encoded_data,
iLBCdec_inst, mode);
/* convert to short */
for (k=0; k<iLBCdec_inst->blockl; k++){
dtmp=decblock[k];
if (dtmp<MIN_SAMPLE)
dtmp=MIN_SAMPLE;
else if (dtmp>MAX_SAMPLE)
dtmp=MAX_SAMPLE;
decoded_data[k] = (short) dtmp;
}
return (iLBCdec_inst->blockl);
}
/*---------------------------------------------------------------*
* Main program to test iLBC encoding and decoding
*
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* Usage:
* exefile_name.exe <infile> <bytefile> <outfile> <channel>
*
* <infile> : Input file, speech for encoder (16-bit pcm file)
* <bytefile> : Bit stream output from the encoder
* <outfile> : Output file, decoded speech (16-bit pcm file)
* <channel> : Bit error file, optional (16-bit)
* 1 - Packet received correctly
* 0 - Packet Lost
*
*--------------------------------------------------------------*/
int main(int argc, char* argv[])
{
/* Runtime statistics */
float starttime;
float runtime;
float outtime;
FILE *ifileid,*efileid,*ofileid, *cfileid;
short data[BLOCKL_MAX];
short encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
int len;
short pli, mode;
int blockcount = 0;
int packetlosscount = 0;
/* Create structs */
iLBC_Enc_Inst_t Enc_Inst;
iLBC_Dec_Inst_t Dec_Inst;
/* get arguments and open files */
if ((argc!=5) && (argc!=6)) {
fprintf(stderr,
"\n*-----------------------------------------------*\n");
fprintf(stderr,
" %s <20,30> input encoded decoded (channel)\n\n",
argv[0]);
fprintf(stderr,
" mode : Frame size for the encoding/decoding\n");
fprintf(stderr,
" 20 - 20 ms\n");
fprintf(stderr,
" 30 - 30 ms\n");
fprintf(stderr,
" input : Speech for encoder (16-bit pcm file)\n");
fprintf(stderr,
" encoded : Encoded bit stream\n");
fprintf(stderr,
" decoded : Decoded speech (16-bit pcm file)\n");
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fprintf(stderr,
" channel : Packet loss pattern, optional (16-bit)\n");
fprintf(stderr,
" 1 - Packet received correctly\n");
fprintf(stderr,
" 0 - Packet Lost\n");
fprintf(stderr,
"*-----------------------------------------------*\n\n");
exit(1);
}
mode=atoi(argv[1]);
if (mode != 20 && mode != 30) {
fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
argv[1]);
exit(2);
}
if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
fprintf(stderr,"Cannot open input file %s\n", argv[2]);
exit(2);}
if ( (efileid=fopen(argv[3],"wb")) == NULL) {
fprintf(stderr, "Cannot open encoded file file %s\n",
argv[3]); exit(1);}
if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
fprintf(stderr, "Cannot open decoded file %s\n",
argv[4]); exit(1);}
if (argc==6) {
if( (cfileid=fopen(argv[5],"rb")) == NULL) {
fprintf(stderr, "Cannot open channel file %s\n",
argv[5]);
exit(1);
}
} else {
cfileid=NULL;
}
/* print info */
fprintf(stderr, "\n");
fprintf(stderr,
"*---------------------------------------------------*\n");
fprintf(stderr,
"* *\n");
fprintf(stderr,
"* iLBC test program *\n");
fprintf(stderr,
"* *\n");
fprintf(stderr,
"* *\n");
fprintf(stderr,
"*---------------------------------------------------*\n");
fprintf(stderr,"\nMode : %2d ms\n", mode);
fprintf(stderr,"Input file : %s\n", argv[2]);
fprintf(stderr,"Encoded file : %s\n", argv[3]);
fprintf(stderr,"Output file : %s\n", argv[4]);
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if (argc==6) {
fprintf(stderr,"Channel file : %s\n", argv[5]);
}
fprintf(stderr,"\n");
/* Initialization */
initEncode(&Enc_Inst, mode);
initDecode(&Dec_Inst, mode, 1);
/* Runtime statistics */
starttime=clock()/(float)CLOCKS_PER_SEC;
/* loop over input blocks */
while (fread(data,sizeof(short),Enc_Inst.blockl,ifileid)==
Enc_Inst.blockl) {
blockcount++;
/* encoding */
fprintf(stderr, "--- Encoding block %i --- ",blockcount);
len=encode(&Enc_Inst, encoded_data, data);
fprintf(stderr, "\r");
/* write byte file */
fwrite(encoded_data, sizeof(unsigned char), len, efileid);
/* get channel data if provided */
if (argc==6) {
if (fread(&pli, sizeof(short), 1, cfileid)) {
if ((pli!=0)&&(pli!=1)) {
fprintf(stderr, "Error in channel file\n");
exit(0);
}
if (pli==0) {
/* Packet loss -> remove info from frame */
memset(encoded_data, 0,
sizeof(short)*ILBCNOOFWORDS_MAX);
packetlosscount++;
}
} else {
fprintf(stderr, "Error. Channel file too short\n");
exit(0);
}
} else {
pli=1;
}
/* decoding */
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fprintf(stderr, "--- Decoding block %i --- ",blockcount);
len=decode(&Dec_Inst, decoded_data, encoded_data, pli);
fprintf(stderr, "\r");
/* write output file */
fwrite(decoded_data,sizeof(short),len,ofileid);
}
/* Runtime statistics */
runtime = (float)(clock()/(float)CLOCKS_PER_SEC-starttime);
outtime = (float)((float)blockcount*(float)mode/1000.0);
printf("\n\nLength of speech file: %.1f s\n", outtime);
printf("Packet loss : %.1f%%\n",
100.0*(float)packetlosscount/(float)blockcount);
printf("Time to run iLBC :");
printf(" %.1f s (%.1f %% of realtime)\n\n", runtime,
(100*runtime/outtime));
/* close files */
fclose(ifileid); fclose(efileid); fclose(ofileid);
if (argc==6) {
fclose(cfileid);
}
return(0);
}
A.2 iLBC_encode.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_encode.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_ILBCENCODE_H
#define __iLBC_ILBCENCODE_H
#include "iLBC_define.h"
short initEncode( /* (o) Number of bytes
encoded */
iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
int mode /* (i) frame size mode */
);
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void iLBC_encode(
unsigned char *bytes, /* (o) encoded data bits iLBC */
float *block, /* (o) speech vector to
encode */
iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
state */
);
#endif
A.3 iLBC_encode.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_encode.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include "iLBC_define.h"
#include "LPCencode.h"
#include "FrameClassify.h"
#include "StateSearchW.h"
#include "StateConstructW.h"
#include "helpfun.h"
#include "constants.h"
#include "packing.h"
#include "iCBSearch.h"
#include "iCBConstruct.h"
#include "hpInput.h"
#include "anaFilter.h"
#include "syntFilter.h"
/*----------------------------------------------------------------*
* Initiation of encoder instance.
*---------------------------------------------------------------*/
short initEncode( /* (o) Number of bytes
encoded */
iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
int mode /* (i) frame size mode */
){
iLBCenc_inst->mode = mode;
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if (mode==30) {
iLBCenc_inst->blockl = BLOCKL_30MS;
iLBCenc_inst->nsub = NSUB_30MS;
iLBCenc_inst->nasub = NASUB_30MS;
iLBCenc_inst->lpc_n = LPC_N_30MS;
iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
/* ULP init */
iLBCenc_inst->ULP_inst=&ULP_30msTbl;
}
else if (mode==20) {
iLBCenc_inst->blockl = BLOCKL_20MS;
iLBCenc_inst->nsub = NSUB_20MS;
iLBCenc_inst->nasub = NASUB_20MS;
iLBCenc_inst->lpc_n = LPC_N_20MS;
iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
/* ULP init */
iLBCenc_inst->ULP_inst=&ULP_20msTbl;
}
else {
exit(2);
}
memset((*iLBCenc_inst).anaMem, 0,
LPC_FILTERORDER*sizeof(float));
memcpy((*iLBCenc_inst).lsfold, lsfmeanTbl,
LPC_FILTERORDER*sizeof(float));
memcpy((*iLBCenc_inst).lsfdeqold, lsfmeanTbl,
LPC_FILTERORDER*sizeof(float));
memset((*iLBCenc_inst).lpc_buffer, 0,
(LPC_LOOKBACK+BLOCKL_MAX)*sizeof(float));
memset((*iLBCenc_inst).hpimem, 0, 4*sizeof(float));
return (iLBCenc_inst->no_of_bytes);
}
/*----------------------------------------------------------------*
* main encoder function
*---------------------------------------------------------------*/
void iLBC_encode(
unsigned char *bytes, /* (o) encoded data bits iLBC */
float *block, /* (o) speech vector to
encode */
iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
state */
){
float data[BLOCKL_MAX];
float residual[BLOCKL_MAX], reverseResidual[BLOCKL_MAX];
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int start, idxForMax, idxVec[STATE_LEN];
float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
int n, k, meml_gotten, Nfor, Nback, i, pos;
int gain_index[CB_NSTAGES*NASUB_MAX],
extra_gain_index[CB_NSTAGES];
int cb_index[CB_NSTAGES*NASUB_MAX],extra_cb_index[CB_NSTAGES];
int lsf_i[LSF_NSPLIT*LPC_N_MAX];
unsigned char *pbytes;
int diff, start_pos, state_first;
float en1, en2;
int index, ulp, firstpart;
int subcount, subframe;
float weightState[LPC_FILTERORDER];
float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
float weightdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
float decresidual[BLOCKL_MAX];
/* high pass filtering of input signal if such is not done
prior to calling this function */
hpInput(block, iLBCenc_inst->blockl,
data, (*iLBCenc_inst).hpimem);
/* otherwise simply copy */
/*memcpy(data,block,iLBCenc_inst->blockl*sizeof(float));*/
/* LPC of hp filtered input data */
LPCencode(syntdenum, weightdenum, lsf_i, data, iLBCenc_inst);
/* inverse filter to get residual */
for (n=0; n<iLBCenc_inst->nsub; n++) {
anaFilter(&data[n*SUBL], &syntdenum[n*(LPC_FILTERORDER+1)],
SUBL, &residual[n*SUBL], iLBCenc_inst->anaMem);
}
/* find state location */
start = FrameClassify(iLBCenc_inst, residual);
/* check if state should be in first or last part of the
two subframes */
diff = STATE_LEN - iLBCenc_inst->state_short_len;
en1 = 0;
index = (start-1)*SUBL;
for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
en1 += residual[index+i]*residual[index+i];
}
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en2 = 0;
index = (start-1)*SUBL+diff;
for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
en2 += residual[index+i]*residual[index+i];
}
if (en1 > en2) {
state_first = 1;
start_pos = (start-1)*SUBL;
} else {
state_first = 0;
start_pos = (start-1)*SUBL + diff;
}
/* scalar quantization of state */
StateSearchW(iLBCenc_inst, &residual[start_pos],
&syntdenum[(start-1)*(LPC_FILTERORDER+1)],
&weightdenum[(start-1)*(LPC_FILTERORDER+1)], &idxForMax,
idxVec, iLBCenc_inst->state_short_len, state_first);
StateConstructW(idxForMax, idxVec,
&syntdenum[(start-1)*(LPC_FILTERORDER+1)],
&decresidual[start_pos], iLBCenc_inst->state_short_len);
/* predictive quantization in state */
if (state_first) { /* put adaptive part in the end */
/* setup memory */
memset(mem, 0,
(CB_MEML-iLBCenc_inst->state_short_len)*sizeof(float));
memcpy(mem+CB_MEML-iLBCenc_inst->state_short_len,
decresidual+start_pos,
iLBCenc_inst->state_short_len*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
/* encode sub-frames */
iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
&residual[start_pos+iLBCenc_inst->state_short_len],
mem+CB_MEML-stMemLTbl,
stMemLTbl, diff, CB_NSTAGES,
&weightdenum[start*(LPC_FILTERORDER+1)],
weightState, 0);
/* construct decoded vector */
iCBConstruct(
&decresidual[start_pos+iLBCenc_inst->state_short_len],
extra_cb_index, extra_gain_index,
mem+CB_MEML-stMemLTbl,
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stMemLTbl, diff, CB_NSTAGES);
}
else { /* put adaptive part in the beginning */
/* create reversed vectors for prediction */
for (k=0; k<diff; k++) {
reverseResidual[k] = residual[(start+1)*SUBL-1
-(k+iLBCenc_inst->state_short_len)];
}
/* setup memory */
meml_gotten = iLBCenc_inst->state_short_len;
for (k=0; k<meml_gotten; k++) {
mem[CB_MEML-1-k] = decresidual[start_pos + k];
}
memset(mem, 0, (CB_MEML-k)*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
/* encode sub-frames */
iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
reverseResidual, mem+CB_MEML-stMemLTbl, stMemLTbl,
diff, CB_NSTAGES,
&weightdenum[(start-1)*(LPC_FILTERORDER+1)],
weightState, 0);
/* construct decoded vector */
iCBConstruct(reverseDecresidual, extra_cb_index,
extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
diff, CB_NSTAGES);
/* get decoded residual from reversed vector */
for (k=0; k<diff; k++) {
decresidual[start_pos-1-k] = reverseDecresidual[k];
}
}
/* counter for predicted sub-frames */
subcount=0;
/* forward prediction of sub-frames */
Nfor = iLBCenc_inst->nsub-start-1;
if ( Nfor > 0 ) {
/* setup memory */
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memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
STATE_LEN*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
/* loop over sub-frames to encode */
for (subframe=0; subframe<Nfor; subframe++) {
/* encode sub-frame */
iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
&residual[(start+1+subframe)*SUBL],
mem+CB_MEML-memLfTbl[subcount],
memLfTbl[subcount], SUBL, CB_NSTAGES,
&weightdenum[(start+1+subframe)*
(LPC_FILTERORDER+1)],
weightState, subcount+1);
/* construct decoded vector */
iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
mem+CB_MEML-memLfTbl[subcount],
memLfTbl[subcount], SUBL, CB_NSTAGES);
/* update memory */
memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
memcpy(mem+CB_MEML-SUBL,
&decresidual[(start+1+subframe)*SUBL],
SUBL*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
subcount++;
}
}
/* backward prediction of sub-frames */
Nback = start-1;
if ( Nback > 0 ) {
/* create reverse order vectors */
for (n=0; n<Nback; n++) {
for (k=0; k<SUBL; k++) {
reverseResidual[n*SUBL+k] =
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residual[(start-1)*SUBL-1-n*SUBL-k];
reverseDecresidual[n*SUBL+k] =
decresidual[(start-1)*SUBL-1-n*SUBL-k];
}
}
/* setup memory */
meml_gotten = SUBL*(iLBCenc_inst->nsub+1-start);
if ( meml_gotten > CB_MEML ) {
meml_gotten=CB_MEML;
}
for (k=0; k<meml_gotten; k++) {
mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
}
memset(mem, 0, (CB_MEML-k)*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
/* loop over sub-frames to encode */
for (subframe=0; subframe<Nback; subframe++) {
/* encode sub-frame */
iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
&reverseResidual[subframe*SUBL],
mem+CB_MEML-memLfTbl[subcount],
memLfTbl[subcount], SUBL, CB_NSTAGES,
&weightdenum[(start-2-subframe)*
(LPC_FILTERORDER+1)],
weightState, subcount+1);
/* construct decoded vector */
iCBConstruct(&reverseDecresidual[subframe*SUBL],
cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
mem+CB_MEML-memLfTbl[subcount],
memLfTbl[subcount], SUBL, CB_NSTAGES);
/* update memory */
memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
memcpy(mem+CB_MEML-SUBL,
&reverseDecresidual[subframe*SUBL],
SUBL*sizeof(float));
memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
subcount++;
}
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/* get decoded residual from reversed vector */
for (i=0; i<SUBL*Nback; i++) {
decresidual[SUBL*Nback - i - 1] =
reverseDecresidual[i];
}
}
/* end encoding part */
/* adjust index */
index_conv_enc(cb_index);
/* pack bytes */
pbytes=bytes;
pos=0;
/* loop over the 3 ULP classes */
for (ulp=0; ulp<3; ulp++) {
/* LSF */
for (k=0; k<LSF_NSPLIT*iLBCenc_inst->lpc_n; k++) {
packsplit(&lsf_i[k], &firstpart, &lsf_i[k],
iLBCenc_inst->ULP_inst->lsf_bits[k][ulp],
iLBCenc_inst->ULP_inst->lsf_bits[k][ulp]+
iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+1]+
iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->lsf_bits[k][ulp], &pos);
}
/* Start block info */
packsplit(&start, &firstpart, &start,
iLBCenc_inst->ULP_inst->start_bits[ulp],
iLBCenc_inst->ULP_inst->start_bits[ulp]+
iLBCenc_inst->ULP_inst->start_bits[ulp+1]+
iLBCenc_inst->ULP_inst->start_bits[ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->start_bits[ulp], &pos);
packsplit(&state_first, &firstpart, &state_first,
iLBCenc_inst->ULP_inst->startfirst_bits[ulp],
iLBCenc_inst->ULP_inst->startfirst_bits[ulp]+
iLBCenc_inst->ULP_inst->startfirst_bits[ulp+1]+
iLBCenc_inst->ULP_inst->startfirst_bits[ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->startfirst_bits[ulp], &pos);
packsplit(&idxForMax, &firstpart, &idxForMax,
iLBCenc_inst->ULP_inst->scale_bits[ulp],
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iLBCenc_inst->ULP_inst->scale_bits[ulp]+
iLBCenc_inst->ULP_inst->scale_bits[ulp+1]+
iLBCenc_inst->ULP_inst->scale_bits[ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->scale_bits[ulp], &pos);
for (k=0; k<iLBCenc_inst->state_short_len; k++) {
packsplit(idxVec+k, &firstpart, idxVec+k,
iLBCenc_inst->ULP_inst->state_bits[ulp],
iLBCenc_inst->ULP_inst->state_bits[ulp]+
iLBCenc_inst->ULP_inst->state_bits[ulp+1]+
iLBCenc_inst->ULP_inst->state_bits[ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->state_bits[ulp], &pos);
}
/* 23/22 (20ms/30ms) sample block */
for (k=0;k<CB_NSTAGES;k++) {
packsplit(extra_cb_index+k, &firstpart,
extra_cb_index+k,
iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp]+
iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+1]+
iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
&pos);
}
for (k=0;k<CB_NSTAGES;k++) {
packsplit(extra_gain_index+k, &firstpart,
extra_gain_index+k,
iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp]+
iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+1]+
iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
&pos);
}
/* The two/four (20ms/30ms) 40 sample sub-blocks */
for (i=0; i<iLBCenc_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
packsplit(cb_index+i*CB_NSTAGES+k, &firstpart,
cb_index+i*CB_NSTAGES+k,
iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
iLBCenc_inst->ULP_inst->cb_index[i][k][ulp]+
iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+1]+
iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
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&pos);
}
}
for (i=0; i<iLBCenc_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
packsplit(gain_index+i*CB_NSTAGES+k, &firstpart,
gain_index+i*CB_NSTAGES+k,
iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp]+
iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+1]+
iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+2]);
dopack( &pbytes, firstpart,
iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
&pos);
}
}
}
/* set the last bit to zero (otherwise the decoder
will treat it as a lost frame) */
dopack( &pbytes, 0, 1, &pos);
}
A.4 iLBC_decode.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_decode.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_ILBCDECODE_H
#define __iLBC_ILBCDECODE_H
#include "iLBC_define.h"
short initDecode( /* (o) Number of decoded
samples */
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
int mode, /* (i) frame size mode */
int use_enhancer /* (i) 1 to use enhancer
0 to run without
enhancer */
);
void iLBC_decode(
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float *decblock, /* (o) decoded signal block */
unsigned char *bytes, /* (i) encoded signal bits */
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
structure */
int mode /* (i) 0: bad packet, PLC,
1: normal */
);
#endif
A.5 iLBC_decode.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_decode.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <stdlib.h>
#include "iLBC_define.h"
#include "StateConstructW.h"
#include "LPCdecode.h"
#include "iCBConstruct.h"
#include "doCPLC.h"
#include "helpfun.h"
#include "constants.h"
#include "packing.h"
#include "string.h"
#include "enhancer.h"
#include "hpOutput.h"
#include "syntFilter.h"
/*----------------------------------------------------------------*
* Initiation of decoder instance.
*---------------------------------------------------------------*/
short initDecode( /* (o) Number of decoded
samples */
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
int mode, /* (i) frame size mode */
int use_enhancer /* (i) 1 to use enhancer
0 to run without
enhancer */
){
int i;
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iLBCdec_inst->mode = mode;
if (mode==30) {
iLBCdec_inst->blockl = BLOCKL_30MS;
iLBCdec_inst->nsub = NSUB_30MS;
iLBCdec_inst->nasub = NASUB_30MS;
iLBCdec_inst->lpc_n = LPC_N_30MS;
iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
/* ULP init */
iLBCdec_inst->ULP_inst=&ULP_30msTbl;
}
else if (mode==20) {
iLBCdec_inst->blockl = BLOCKL_20MS;
iLBCdec_inst->nsub = NSUB_20MS;
iLBCdec_inst->nasub = NASUB_20MS;
iLBCdec_inst->lpc_n = LPC_N_20MS;
iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
/* ULP init */
iLBCdec_inst->ULP_inst=&ULP_20msTbl;
}
else {
exit(2);
}
memset(iLBCdec_inst->syntMem, 0,
LPC_FILTERORDER*sizeof(float));
memcpy((*iLBCdec_inst).lsfdeqold, lsfmeanTbl,
LPC_FILTERORDER*sizeof(float));
memset(iLBCdec_inst->old_syntdenum, 0,
((LPC_FILTERORDER + 1)*NSUB_MAX)*sizeof(float));
for (i=0; i<NSUB_MAX; i++)
iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]=1.0;
iLBCdec_inst->last_lag = 20;
iLBCdec_inst->prevLag = 120;
iLBCdec_inst->per = 0.0;
iLBCdec_inst->consPLICount = 0;
iLBCdec_inst->prevPLI = 0;
iLBCdec_inst->prevLpc[0] = 1.0;
memset(iLBCdec_inst->prevLpc+1,0,
LPC_FILTERORDER*sizeof(float));
memset(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX*sizeof(float));
iLBCdec_inst->seed=777;
memset(iLBCdec_inst->hpomem, 0, 4*sizeof(float));
iLBCdec_inst->use_enhancer = use_enhancer;
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memset(iLBCdec_inst->enh_buf, 0, ENH_BUFL*sizeof(float));
for (i=0;i<ENH_NBLOCKS_TOT;i++)
iLBCdec_inst->enh_period[i]=(float)40.0;
iLBCdec_inst->prev_enh_pl = 0;
return (iLBCdec_inst->blockl);
}
/*----------------------------------------------------------------*
* frame residual decoder function (subrutine to iLBC_decode)
*---------------------------------------------------------------*/
void Decode(
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
structure */
float *decresidual, /* (o) decoded residual frame */
int start, /* (i) location of start
state */
int idxForMax, /* (i) codebook index for the
maximum value */
int *idxVec, /* (i) codebook indexes for the
samples in the start
state */
float *syntdenum, /* (i) the decoded synthesis
filter coefficients */
int *cb_index, /* (i) the indexes for the
adaptive codebook */
int *gain_index, /* (i) the indexes for the
corresponding gains */
int *extra_cb_index, /* (i) the indexes for the
adaptive codebook part
of start state */
int *extra_gain_index, /* (i) the indexes for the
corresponding gains */
int state_first /* (i) 1 if non adaptive part
of start state comes
first 0 if that part
comes last */
){
float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
int k, meml_gotten, Nfor, Nback, i;
int diff, start_pos;
int subcount, subframe;
diff = STATE_LEN - iLBCdec_inst->state_short_len;
if (state_first == 1) {
start_pos = (start-1)*SUBL;
} else {
start_pos = (start-1)*SUBL + diff;
}
/* decode scalar part of start state */
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StateConstructW(idxForMax, idxVec,
&syntdenum[(start-1)*(LPC_FILTERORDER+1)],
&decresidual[start_pos], iLBCdec_inst->state_short_len);
if (state_first) { /* put adaptive part in the end */
/* setup memory */
memset(mem, 0,
(CB_MEML-iLBCdec_inst->state_short_len)*sizeof(float));
memcpy(mem+CB_MEML-iLBCdec_inst->state_short_len,
decresidual+start_pos,
iLBCdec_inst->state_short_len*sizeof(float));
/* construct decoded vector */
iCBConstruct(
&decresidual[start_pos+iLBCdec_inst->state_short_len],
extra_cb_index, extra_gain_index, mem+CB_MEML-stMemLTbl,
stMemLTbl, diff, CB_NSTAGES);
}
else {/* put adaptive part in the beginning */
/* create reversed vectors for prediction */
for (k=0; k<diff; k++) {
reverseDecresidual[k] =
decresidual[(start+1)*SUBL-1-
(k+iLBCdec_inst->state_short_len)];
}
/* setup memory */
meml_gotten = iLBCdec_inst->state_short_len;
for (k=0; k<meml_gotten; k++){
mem[CB_MEML-1-k] = decresidual[start_pos + k];
}
memset(mem, 0, (CB_MEML-k)*sizeof(float));
/* construct decoded vector */
iCBConstruct(reverseDecresidual, extra_cb_index,
extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
diff, CB_NSTAGES);
/* get decoded residual from reversed vector */
for (k=0; k<diff; k++) {
decresidual[start_pos-1-k] = reverseDecresidual[k];
}
}
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/* counter for predicted sub-frames */
subcount=0;
/* forward prediction of sub-frames */
Nfor = iLBCdec_inst->nsub-start-1;
if ( Nfor > 0 ){
/* setup memory */
memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
STATE_LEN*sizeof(float));
/* loop over sub-frames to encode */
for (subframe=0; subframe<Nfor; subframe++) {
/* construct decoded vector */
iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
mem+CB_MEML-memLfTbl[subcount],
memLfTbl[subcount], SUBL, CB_NSTAGES);
/* update memory */
memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
memcpy(mem+CB_MEML-SUBL,
&decresidual[(start+1+subframe)*SUBL],
SUBL*sizeof(float));
subcount++;
}
}
/* backward prediction of sub-frames */
Nback = start-1;
if ( Nback > 0 ) {
/* setup memory */
meml_gotten = SUBL*(iLBCdec_inst->nsub+1-start);
if ( meml_gotten > CB_MEML ) {
meml_gotten=CB_MEML;
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}
for (k=0; k<meml_gotten; k++) {
mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
}
memset(mem, 0, (CB_MEML-k)*sizeof(float));
/* loop over subframes to decode */
for (subframe=0; subframe<Nback; subframe++) {
/* construct decoded vector */
iCBConstruct(&reverseDecresidual[subframe*SUBL],
cb_index+subcount*CB_NSTAGES,
gain_index+subcount*CB_NSTAGES,
mem+CB_MEML-memLfTbl[subcount], memLfTbl[subcount],
SUBL, CB_NSTAGES);
/* update memory */
memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
memcpy(mem+CB_MEML-SUBL,
&reverseDecresidual[subframe*SUBL],
SUBL*sizeof(float));
subcount++;
}
/* get decoded residual from reversed vector */
for (i=0; i<SUBL*Nback; i++)
decresidual[SUBL*Nback - i - 1] =
reverseDecresidual[i];
}
}
/*----------------------------------------------------------------*
* main decoder function
*---------------------------------------------------------------*/
void iLBC_decode(
float *decblock, /* (o) decoded signal block */
unsigned char *bytes, /* (i) encoded signal bits */
iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
structure */
int mode /* (i) 0: bad packet, PLC,
1: normal */
){
float data[BLOCKL_MAX];
float lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
float PLCresidual[BLOCKL_MAX], PLClpc[LPC_FILTERORDER + 1];
float zeros[BLOCKL_MAX], one[LPC_FILTERORDER + 1];
int k, i, start, idxForMax, pos, lastpart, ulp;
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int lag, ilag;
float cc, maxcc;
int idxVec[STATE_LEN];
int check;
int gain_index[NASUB_MAX*CB_NSTAGES],
extra_gain_index[CB_NSTAGES];
int cb_index[CB_NSTAGES*NASUB_MAX], extra_cb_index[CB_NSTAGES];
int lsf_i[LSF_NSPLIT*LPC_N_MAX];
int state_first;
int last_bit;
unsigned char *pbytes;
float weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
int order_plus_one;
float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
float decresidual[BLOCKL_MAX];
if (mode>0) { /* the data are good */
/* decode data */
pbytes=bytes;
pos=0;
/* Set everything to zero before decoding */
for (k=0; k<LSF_NSPLIT*LPC_N_MAX; k++) {
lsf_i[k]=0;
}
start=0;
state_first=0;
idxForMax=0;
for (k=0; k<iLBCdec_inst->state_short_len; k++) {
idxVec[k]=0;
}
for (k=0; k<CB_NSTAGES; k++) {
extra_cb_index[k]=0;
}
for (k=0; k<CB_NSTAGES; k++) {
extra_gain_index[k]=0;
}
for (i=0; i<iLBCdec_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
cb_index[i*CB_NSTAGES+k]=0;
}
}
for (i=0; i<iLBCdec_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
gain_index[i*CB_NSTAGES+k]=0;
}
}
/* loop over ULP classes */
for (ulp=0; ulp<3; ulp++) {
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/* LSF */
for (k=0; k<LSF_NSPLIT*iLBCdec_inst->lpc_n; k++){
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->lsf_bits[k][ulp], &pos);
packcombine(&lsf_i[k], lastpart,
iLBCdec_inst->ULP_inst->lsf_bits[k][ulp]);
}
/* Start block info */
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->start_bits[ulp], &pos);
packcombine(&start, lastpart,
iLBCdec_inst->ULP_inst->start_bits[ulp]);
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->startfirst_bits[ulp], &pos);
packcombine(&state_first, lastpart,
iLBCdec_inst->ULP_inst->startfirst_bits[ulp]);
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->scale_bits[ulp], &pos);
packcombine(&idxForMax, lastpart,
iLBCdec_inst->ULP_inst->scale_bits[ulp]);
for (k=0; k<iLBCdec_inst->state_short_len; k++) {
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->state_bits[ulp], &pos);
packcombine(idxVec+k, lastpart,
iLBCdec_inst->ULP_inst->state_bits[ulp]);
}
/* 23/22 (20ms/30ms) sample block */
for (k=0; k<CB_NSTAGES; k++) {
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp],
&pos);
packcombine(extra_cb_index+k, lastpart,
iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp]);
}
for (k=0; k<CB_NSTAGES; k++) {
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp],
&pos);
packcombine(extra_gain_index+k, lastpart,
iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp]);
}
/* The two/four (20ms/30ms) 40 sample sub-blocks */
for (i=0; i<iLBCdec_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
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unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->cb_index[i][k][ulp],
&pos);
packcombine(cb_index+i*CB_NSTAGES+k, lastpart,
iLBCdec_inst->ULP_inst->cb_index[i][k][ulp]);
}
}
for (i=0; i<iLBCdec_inst->nasub; i++) {
for (k=0; k<CB_NSTAGES; k++) {
unpack( &pbytes, &lastpart,
iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp],
&pos);
packcombine(gain_index+i*CB_NSTAGES+k, lastpart,
iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp]);
}
}
}
/* Extract last bit. If it is 1 this indicates an
empty/lost frame */
unpack( &pbytes, &last_bit, 1, &pos);
/* Check for bit errors or empty/lost frames */
if (start<1)
mode = 0;
if (iLBCdec_inst->mode==20 && start>3)
mode = 0;
if (iLBCdec_inst->mode==30 && start>5)
mode = 0;
if (last_bit==1)
mode = 0;
if (mode==1) { /* No bit errors was detected,
continue decoding */
/* adjust index */
index_conv_dec(cb_index);
/* decode the lsf */
SimplelsfDEQ(lsfdeq, lsf_i, iLBCdec_inst->lpc_n);
check=LSF_check(lsfdeq, LPC_FILTERORDER,
iLBCdec_inst->lpc_n);
DecoderInterpolateLSF(syntdenum, weightdenum,
lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
Decode(iLBCdec_inst, decresidual, start, idxForMax,
idxVec, syntdenum, cb_index, gain_index,
extra_cb_index, extra_gain_index,
state_first);
/* preparing the plc for a future loss! */
doThePLC(PLCresidual, PLClpc, 0, decresidual,
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syntdenum +
(LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
(*iLBCdec_inst).last_lag, iLBCdec_inst);
memcpy(decresidual, PLCresidual,
iLBCdec_inst->blockl*sizeof(float));
}
}
if (mode == 0) {
/* the data is bad (either a PLC call
* was made or a severe bit error was detected)
*/
/* packet loss conceal */
memset(zeros, 0, BLOCKL_MAX*sizeof(float));
one[0] = 1;
memset(one+1, 0, LPC_FILTERORDER*sizeof(float));
start=0;
doThePLC(PLCresidual, PLClpc, 1, zeros, one,
(*iLBCdec_inst).last_lag, iLBCdec_inst);
memcpy(decresidual, PLCresidual,
iLBCdec_inst->blockl*sizeof(float));
order_plus_one = LPC_FILTERORDER + 1;
for (i = 0; i < iLBCdec_inst->nsub; i++) {
memcpy(syntdenum+(i*order_plus_one), PLClpc,
order_plus_one*sizeof(float));
}
}
if (iLBCdec_inst->use_enhancer == 1) {
/* post filtering */
iLBCdec_inst->last_lag =
enhancerInterface(data, decresidual, iLBCdec_inst);
/* synthesis filtering */
if (iLBCdec_inst->mode==20) {
/* Enhancer has 40 samples delay */
i=0;
syntFilter(data + i*SUBL,
iLBCdec_inst->old_syntdenum +
(i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
SUBL, iLBCdec_inst->syntMem);
for (i=1; i < iLBCdec_inst->nsub; i++) {
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syntFilter(data + i*SUBL,
syntdenum + (i-1)*(LPC_FILTERORDER+1),
SUBL, iLBCdec_inst->syntMem);
}
} else if (iLBCdec_inst->mode==30) {
/* Enhancer has 80 samples delay */
for (i=0; i < 2; i++) {
syntFilter(data + i*SUBL,
iLBCdec_inst->old_syntdenum +
(i+iLBCdec_inst->nsub-2)*(LPC_FILTERORDER+1),
SUBL, iLBCdec_inst->syntMem);
}
for (i=2; i < iLBCdec_inst->nsub; i++) {
syntFilter(data + i*SUBL,
syntdenum + (i-2)*(LPC_FILTERORDER+1), SUBL,
iLBCdec_inst->syntMem);
}
}
} else {
/* Find last lag */
lag = 20;
maxcc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
&decresidual[BLOCKL_MAX-ENH_BLOCKL-lag], ENH_BLOCKL);
for (ilag=21; ilag<120; ilag++) {
cc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
&decresidual[BLOCKL_MAX-ENH_BLOCKL-ilag],
ENH_BLOCKL);
if (cc > maxcc) {
maxcc = cc;
lag = ilag;
}
}
iLBCdec_inst->last_lag = lag;
/* copy data and run synthesis filter */
memcpy(data, decresidual,
iLBCdec_inst->blockl*sizeof(float));
for (i=0; i < iLBCdec_inst->nsub; i++) {
syntFilter(data + i*SUBL,
syntdenum + i*(LPC_FILTERORDER+1), SUBL,
iLBCdec_inst->syntMem);
}
}
/* high pass filtering on output if desired, otherwise
copy to out */
hpOutput(data, iLBCdec_inst->blockl,
decblock,iLBCdec_inst->hpomem);
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/* memcpy(decblock,data,iLBCdec_inst->blockl*sizeof(float));*/
memcpy(iLBCdec_inst->old_syntdenum, syntdenum,
iLBCdec_inst->nsub*(LPC_FILTERORDER+1)*sizeof(float));
iLBCdec_inst->prev_enh_pl=0;
if (mode==0) { /* PLC was used */
iLBCdec_inst->prev_enh_pl=1;
}
}
A.6 iLBC_define.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iLBC_define.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <string.h>
#ifndef __iLBC_ILBCDEFINE_H
#define __iLBC_ILBCDEFINE_H
/* general codec settings */
#define FS (float)8000.0
#define BLOCKL_20MS 160
#define BLOCKL_30MS 240
#define BLOCKL_MAX 240
#define NSUB_20MS 4
#define NSUB_30MS 6
#define NSUB_MAX 6
#define NASUB_20MS 2
#define NASUB_30MS 4
#define NASUB_MAX 4
#define SUBL 40
#define STATE_LEN 80
#define STATE_SHORT_LEN_30MS 58
#define STATE_SHORT_LEN_20MS 57
/* LPC settings */
#define LPC_FILTERORDER 10
#define LPC_CHIRP_SYNTDENUM (float)0.9025
#define LPC_CHIRP_WEIGHTDENUM (float)0.4222
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#define LPC_LOOKBACK 60
#define LPC_N_20MS 1
#define LPC_N_30MS 2
#define LPC_N_MAX 2
#define LPC_ASYMDIFF 20
#define LPC_BW (float)60.0
#define LPC_WN (float)1.0001
#define LSF_NSPLIT 3
#define LSF_NUMBER_OF_STEPS 4
#define LPC_HALFORDER (LPC_FILTERORDER/2)
/* cb settings */
#define CB_NSTAGES 3
#define CB_EXPAND 2
#define CB_MEML 147
#define CB_FILTERLEN 2*4
#define CB_HALFFILTERLEN 4
#define CB_RESRANGE 34
#define CB_MAXGAIN (float)1.3
/* enhancer */
#define ENH_BLOCKL 80 /* block length */
#define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
#define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
in said second sequence */
#define ENH_SLOP 2 /* max difference estimated and
correct pitch period */
#define ENH_PLOCSL 20 /* pitch-estimates and pitch-
locations buffer length */
#define ENH_OVERHANG 2
#define ENH_UPS0 4 /* upsampling rate */
#define ENH_FL0 3 /* 2*FLO+1 is the length of
each filter */
#define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
#define ENH_CORRDIM (2*ENH_SLOP+1)
#define ENH_NBLOCKS (BLOCKL_MAX/ENH_BLOCKL)
#define ENH_NBLOCKS_EXTRA 5
#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS +
ENH_NBLOCKS_EXTRA */
#define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
#define ENH_ALPHA0 (float)0.05
/* Down sampling */
#define FILTERORDER_DS 7
#define DELAY_DS 3
#define FACTOR_DS 2
/* bit stream defs */
#define NO_OF_BYTES_20MS 38
#define NO_OF_BYTES_30MS 50
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#define NO_OF_WORDS_20MS 19
#define NO_OF_WORDS_30MS 25
#define STATE_BITS 3
#define BYTE_LEN 8
#define ULP_CLASSES 3
/* help parameters */
#define FLOAT_MAX (float)1.0e37
#define EPS (float)2.220446049250313e-016
#define PI (float)3.14159265358979323846
#define MIN_SAMPLE -32768
#define MAX_SAMPLE 32767
#define TWO_PI (float)6.283185307
#define PI2 (float)0.159154943
/* type definition encoder instance */
typedef struct iLBC_ULP_Inst_t_ {
int lsf_bits[6][ULP_CLASSES+2];
int start_bits[ULP_CLASSES+2];
int startfirst_bits[ULP_CLASSES+2];
int scale_bits[ULP_CLASSES+2];
int state_bits[ULP_CLASSES+2];
int extra_cb_index[CB_NSTAGES][ULP_CLASSES+2];
int extra_cb_gain[CB_NSTAGES][ULP_CLASSES+2];
int cb_index[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
int cb_gain[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
} iLBC_ULP_Inst_t;
/* type definition encoder instance */
typedef struct iLBC_Enc_Inst_t_ {
/* flag for frame size mode */
int mode;
/* basic parameters for different frame sizes */
int blockl;
int nsub;
int nasub;
int no_of_bytes, no_of_words;
int lpc_n;
int state_short_len;
const iLBC_ULP_Inst_t *ULP_inst;
/* analysis filter state */
float anaMem[LPC_FILTERORDER];
/* old lsf parameters for interpolation */
float lsfold[LPC_FILTERORDER];
float lsfdeqold[LPC_FILTERORDER];
/* signal buffer for LP analysis */
float lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
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/* state of input HP filter */
float hpimem[4];
} iLBC_Enc_Inst_t;
/* type definition decoder instance */
typedef struct iLBC_Dec_Inst_t_ {
/* flag for frame size mode */
int mode;
/* basic parameters for different frame sizes */
int blockl;
int nsub;
int nasub;
int no_of_bytes, no_of_words;
int lpc_n;
int state_short_len;
const iLBC_ULP_Inst_t *ULP_inst;
/* synthesis filter state */
float syntMem[LPC_FILTERORDER];
/* old LSF for interpolation */
float lsfdeqold[LPC_FILTERORDER];
/* pitch lag estimated in enhancer and used in PLC */
int last_lag;
/* PLC state information */
int prevLag, consPLICount, prevPLI, prev_enh_pl;
float prevLpc[LPC_FILTERORDER+1];
float prevResidual[NSUB_MAX*SUBL];
float per;
unsigned long seed;
/* previous synthesis filter parameters */
float old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
/* state of output HP filter */
float hpomem[4];
/* enhancer state information */
int use_enhancer;
float enh_buf[ENH_BUFL];
float enh_period[ENH_NBLOCKS_TOT];
} iLBC_Dec_Inst_t;
#endif
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A.7 constants.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
constants.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_CONSTANTS_H
#define __iLBC_CONSTANTS_H
#include "iLBC_define.h"
/* ULP bit allocation */
extern const iLBC_ULP_Inst_t ULP_20msTbl;
extern const iLBC_ULP_Inst_t ULP_30msTbl;
/* high pass filters */
extern float hpi_zero_coefsTbl[];
extern float hpi_pole_coefsTbl[];
extern float hpo_zero_coefsTbl[];
extern float hpo_pole_coefsTbl[];
/* low pass filters */
extern float lpFilt_coefsTbl[];
/* LPC analysis and quantization */
extern float lpc_winTbl[];
extern float lpc_asymwinTbl[];
extern float lpc_lagwinTbl[];
extern float lsfCbTbl[];
extern float lsfmeanTbl[];
extern int dim_lsfCbTbl[];
extern int size_lsfCbTbl[];
extern float lsf_weightTbl_30ms[];
extern float lsf_weightTbl_20ms[];
/* state quantization tables */
extern float state_sq3Tbl[];
extern float state_frgqTbl[];
/* gain quantization tables */
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extern float gain_sq3Tbl[];
extern float gain_sq4Tbl[];
extern float gain_sq5Tbl[];
/* adaptive codebook definitions */
extern int search_rangeTbl[5][CB_NSTAGES];
extern int memLfTbl[];
extern int stMemLTbl;
extern float cbfiltersTbl[CB_FILTERLEN];
/* enhancer definitions */
extern float polyphaserTbl[];
extern float enh_plocsTbl[];
#endif
A.8 constants.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
constants.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "iLBC_define.h"
/* ULP bit allocation */
/* 20 ms frame */
const iLBC_ULP_Inst_t ULP_20msTbl = {
/* LSF */
{ {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
/* Start state location, gain and samples */
{2,0,0,0,0},
{1,0,0,0,0},
{6,0,0,0,0},
{0,1,2,0,0},
/* extra CB index and extra CB gain */
{{6,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
{{2,0,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
/* CB index and CB gain */
{ {{7,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
{{0,0,8,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
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{{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
{{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}},
{ {{1,2,2,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
{{1,1,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
{{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
{{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}}
};
/* 30 ms frame */
const iLBC_ULP_Inst_t ULP_30msTbl = {
/* LSF */
{ {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
{6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0}},
/* Start state location, gain and samples */
{3,0,0,0,0},
{1,0,0,0,0},
{6,0,0,0,0},
{0,1,2,0,0},
/* extra CB index and extra CB gain */
{{4,2,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
{{1,1,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
/* CB index and CB gain */
{ {{6,1,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
{{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
{{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
{{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}}},
{ {{1,2,2,0,0}, {1,2,1,0,0}, {0,0,3,0,0}},
{{0,2,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
{{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}},
{{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}}}
};
/* HP Filters */
float hpi_zero_coefsTbl[3] = {
(float)0.92727436, (float)-1.8544941, (float)0.92727436
};
float hpi_pole_coefsTbl[3] = {
(float)1.0, (float)-1.9059465, (float)0.9114024
};
float hpo_zero_coefsTbl[3] = {
(float)0.93980581, (float)-1.8795834, (float)0.93980581
};
float hpo_pole_coefsTbl[3] = {
(float)1.0, (float)-1.9330735, (float)0.93589199
};
/* LP Filter */
float lpFilt_coefsTbl[FILTERORDER_DS]={
(float)-0.066650, (float)0.125000, (float)0.316650,
(float)0.414063, (float)0.316650,
(float)0.125000, (float)-0.066650
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};
/* State quantization tables */
float state_sq3Tbl[8] = {
(float)-3.719849, (float)-2.177490, (float)-1.130005,
(float)-0.309692, (float)0.444214, (float)1.329712,
(float)2.436279, (float)3.983887
};
float state_frgqTbl[64] = {
(float)1.000085, (float)1.071695, (float)1.140395,
(float)1.206868, (float)1.277188, (float)1.351503,
(float)1.429380, (float)1.500727, (float)1.569049,
(float)1.639599, (float)1.707071, (float)1.781531,
(float)1.840799, (float)1.901550, (float)1.956695,
(float)2.006750, (float)2.055474, (float)2.102787,
(float)2.142819, (float)2.183592, (float)2.217962,
(float)2.257177, (float)2.295739, (float)2.332967,
(float)2.369248, (float)2.402792, (float)2.435080,
(float)2.468598, (float)2.503394, (float)2.539284,
(float)2.572944, (float)2.605036, (float)2.636331,
(float)2.668939, (float)2.698780, (float)2.729101,
(float)2.759786, (float)2.789834, (float)2.818679,
(float)2.848074, (float)2.877470, (float)2.906899,
(float)2.936655, (float)2.967804, (float)3.000115,
(float)3.033367, (float)3.066355, (float)3.104231,
(float)3.141499, (float)3.183012, (float)3.222952,
(float)3.265433, (float)3.308441, (float)3.350823,
(float)3.395275, (float)3.442793, (float)3.490801,
(float)3.542514, (float)3.604064, (float)3.666050,
(float)3.740994, (float)3.830749, (float)3.938770,
(float)4.101764
};
/* CB tables */
int search_rangeTbl[5][CB_NSTAGES]={{58,58,58}, {108,44,44},
{108,108,108}, {108,108,108}, {108,108,108}};
int stMemLTbl=85;
int memLfTbl[NASUB_MAX]={147,147,147,147};
/* expansion filter(s) */
float cbfiltersTbl[CB_FILTERLEN]={
(float)-0.034180, (float)0.108887, (float)-0.184326,
(float)0.806152, (float)0.713379, (float)-0.144043,
(float)0.083740, (float)-0.033691
};
/* Gain Quantization */
float gain_sq3Tbl[8]={
(float)-1.000000, (float)-0.659973, (float)-0.330017,
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(float)0.000000, (float)0.250000, (float)0.500000,
(float)0.750000, (float)1.00000};
float gain_sq4Tbl[16]={
(float)-1.049988, (float)-0.900024, (float)-0.750000,
(float)-0.599976, (float)-0.450012, (float)-0.299988,
(float)-0.150024, (float)0.000000, (float)0.150024,
(float)0.299988, (float)0.450012, (float)0.599976,
(float)0.750000, (float)0.900024, (float)1.049988,
(float)1.200012};
float gain_sq5Tbl[32]={
(float)0.037476, (float)0.075012, (float)0.112488,
(float)0.150024, (float)0.187500, (float)0.224976,
(float)0.262512, (float)0.299988, (float)0.337524,
(float)0.375000, (float)0.412476, (float)0.450012,
(float)0.487488, (float)0.525024, (float)0.562500,
(float)0.599976, (float)0.637512, (float)0.674988,
(float)0.712524, (float)0.750000, (float)0.787476,
(float)0.825012, (float)0.862488, (float)0.900024,
(float)0.937500, (float)0.974976, (float)1.012512,
(float)1.049988, (float)1.087524, (float)1.125000,
(float)1.162476, (float)1.200012};
/* Enhancer - Upsamling a factor 4 (ENH_UPS0 = 4) */
float polyphaserTbl[ENH_UPS0*(2*ENH_FL0+1)]={
(float)0.000000, (float)0.000000, (float)0.000000,
(float)1.000000,
(float)0.000000, (float)0.000000, (float)0.000000,
(float)0.015625, (float)-0.076904, (float)0.288330,
(float)0.862061,
(float)-0.106445, (float)0.018799, (float)-0.015625,
(float)0.023682, (float)-0.124268, (float)0.601563,
(float)0.601563,
(float)-0.124268, (float)0.023682, (float)-0.023682,
(float)0.018799, (float)-0.106445, (float)0.862061,
(float)0.288330,
(float)-0.076904, (float)0.015625, (float)-0.018799};
float enh_plocsTbl[ENH_NBLOCKS_TOT] = {(float)40.0, (float)120.0,
(float)200.0, (float)280.0, (float)360.0,
(float)440.0, (float)520.0, (float)600.0};
/* LPC analysis and quantization */
int dim_lsfCbTbl[LSF_NSPLIT] = {3, 3, 4};
int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
float lsfmeanTbl[LPC_FILTERORDER] = {
(float)0.281738, (float)0.445801, (float)0.663330,
(float)0.962524, (float)1.251831, (float)1.533081,
(float)1.850586, (float)2.137817, (float)2.481445,
(float)2.777344};
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float lsf_weightTbl_30ms[6] = {(float)(1.0/2.0), (float)1.0,
(float)(2.0/3.0),
(float)(1.0/3.0), (float)0.0, (float)0.0};
float lsf_weightTbl_20ms[4] = {(float)(3.0/4.0), (float)(2.0/4.0),
(float)(1.0/4.0), (float)(0.0)};
/* Hanning LPC window */
float lpc_winTbl[BLOCKL_MAX]={
(float)0.000183, (float)0.000671, (float)0.001526,
(float)0.002716, (float)0.004242, (float)0.006104,
(float)0.008301, (float)0.010834, (float)0.013702,
(float)0.016907, (float)0.020416, (float)0.024261,
(float)0.028442, (float)0.032928, (float)0.037750,
(float)0.042877, (float)0.048309, (float)0.054047,
(float)0.060089, (float)0.066437, (float)0.073090,
(float)0.080017, (float)0.087219, (float)0.094727,
(float)0.102509, (float)0.110535, (float)0.118835,
(float)0.127411, (float)0.136230, (float)0.145294,
(float)0.154602, (float)0.164154, (float)0.173920,
(float)0.183899, (float)0.194122, (float)0.204529,
(float)0.215149, (float)0.225952, (float)0.236938,
(float)0.248108, (float)0.259460, (float)0.270966,
(float)0.282654, (float)0.294464, (float)0.306396,
(float)0.318481, (float)0.330688, (float)0.343018,
(float)0.355438, (float)0.367981, (float)0.380585,
(float)0.393280, (float)0.406067, (float)0.418884,
(float)0.431763, (float)0.444702, (float)0.457672,
(float)0.470673, (float)0.483704, (float)0.496735,
(float)0.509766, (float)0.522797, (float)0.535828,
(float)0.548798, (float)0.561768, (float)0.574677,
(float)0.587524, (float)0.600342, (float)0.613068,
(float)0.625732, (float)0.638306, (float)0.650787,
(float)0.663147, (float)0.675415, (float)0.687561,
(float)0.699585, (float)0.711487, (float)0.723206,
(float)0.734802, (float)0.746216, (float)0.757477,
(float)0.768585, (float)0.779480, (float)0.790192,
(float)0.800720, (float)0.811005, (float)0.821106,
(float)0.830994, (float)0.840668, (float)0.850067,
(float)0.859253, (float)0.868225, (float)0.876892,
(float)0.885345, (float)0.893524, (float)0.901428,
(float)0.909058, (float)0.916412, (float)0.923492,
(float)0.930267, (float)0.936768, (float)0.942963,
(float)0.948853, (float)0.954437, (float)0.959717,
(float)0.964691, (float)0.969360, (float)0.973694,
(float)0.977692, (float)0.981384, (float)0.984741,
(float)0.987762, (float)0.990479, (float)0.992828,
(float)0.994873, (float)0.996552, (float)0.997925,
(float)0.998932, (float)0.999603, (float)0.999969,
(float)0.999969, (float)0.999603, (float)0.998932,
(float)0.997925, (float)0.996552, (float)0.994873,
(float)0.992828, (float)0.990479, (float)0.987762,
(float)0.984741, (float)0.981384, (float)0.977692,
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(float)0.973694, (float)0.969360, (float)0.964691,
(float)0.959717, (float)0.954437, (float)0.948853,
(float)0.942963, (float)0.936768, (float)0.930267,
(float)0.923492, (float)0.916412, (float)0.909058,
(float)0.901428, (float)0.893524, (float)0.885345,
(float)0.876892, (float)0.868225, (float)0.859253,
(float)0.850067, (float)0.840668, (float)0.830994,
(float)0.821106, (float)0.811005, (float)0.800720,
(float)0.790192, (float)0.779480, (float)0.768585,
(float)0.757477, (float)0.746216, (float)0.734802,
(float)0.723206, (float)0.711487, (float)0.699585,
(float)0.687561, (float)0.675415, (float)0.663147,
(float)0.650787, (float)0.638306, (float)0.625732,
(float)0.613068, (float)0.600342, (float)0.587524,
(float)0.574677, (float)0.561768, (float)0.548798,
(float)0.535828, (float)0.522797, (float)0.509766,
(float)0.496735, (float)0.483704, (float)0.470673,
(float)0.457672, (float)0.444702, (float)0.431763,
(float)0.418884, (float)0.406067, (float)0.393280,
(float)0.380585, (float)0.367981, (float)0.355438,
(float)0.343018, (float)0.330688, (float)0.318481,
(float)0.306396, (float)0.294464, (float)0.282654,
(float)0.270966, (float)0.259460, (float)0.248108,
(float)0.236938, (float)0.225952, (float)0.215149,
(float)0.204529, (float)0.194122, (float)0.183899,
(float)0.173920, (float)0.164154, (float)0.154602,
(float)0.145294, (float)0.136230, (float)0.127411,
(float)0.118835, (float)0.110535, (float)0.102509,
(float)0.094727, (float)0.087219, (float)0.080017,
(float)0.073090, (float)0.066437, (float)0.060089,
(float)0.054047, (float)0.048309, (float)0.042877,
(float)0.037750, (float)0.032928, (float)0.028442,
(float)0.024261, (float)0.020416, (float)0.016907,
(float)0.013702, (float)0.010834, (float)0.008301,
(float)0.006104, (float)0.004242, (float)0.002716,
(float)0.001526, (float)0.000671, (float)0.000183
};
/* Asymmetric LPC window */
float lpc_asymwinTbl[BLOCKL_MAX]={
(float)0.000061, (float)0.000214, (float)0.000458,
(float)0.000824, (float)0.001282, (float)0.001831,
(float)0.002472, (float)0.003235, (float)0.004120,
(float)0.005066, (float)0.006134, (float)0.007294,
(float)0.008545, (float)0.009918, (float)0.011383,
(float)0.012939, (float)0.014587, (float)0.016357,
(float)0.018219, (float)0.020172, (float)0.022217,
(float)0.024353, (float)0.026611, (float)0.028961,
(float)0.031372, (float)0.033905, (float)0.036530,
(float)0.039276, (float)0.042084, (float)0.044983,
(float)0.047974, (float)0.051086, (float)0.054260,
(float)0.057526, (float)0.060883, (float)0.064331,
(float)0.067871, (float)0.071503, (float)0.075226,
(float)0.079010, (float)0.082916, (float)0.086884,
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(float)0.090942, (float)0.095062, (float)0.099304,
(float)0.103607, (float)0.107971, (float)0.112427,
(float)0.116974, (float)0.121582, (float)0.126282,
(float)0.131073, (float)0.135895, (float)0.140839,
(float)0.145813, (float)0.150879, (float)0.156006,
(float)0.161224, (float)0.166504, (float)0.171844,
(float)0.177246, (float)0.182709, (float)0.188263,
(float)0.193848, (float)0.199524, (float)0.205231,
(float)0.211029, (float)0.216858, (float)0.222778,
(float)0.228729, (float)0.234741, (float)0.240814,
(float)0.246918, (float)0.253082, (float)0.259308,
(float)0.265564, (float)0.271881, (float)0.278259,
(float)0.284668, (float)0.291107, (float)0.297607,
(float)0.304138, (float)0.310730, (float)0.317322,
(float)0.323975, (float)0.330658, (float)0.337372,
(float)0.344147, (float)0.350922, (float)0.357727,
(float)0.364594, (float)0.371460, (float)0.378357,
(float)0.385284, (float)0.392212, (float)0.399170,
(float)0.406158, (float)0.413177, (float)0.420197,
(float)0.427246, (float)0.434296, (float)0.441376,
(float)0.448456, (float)0.455536, (float)0.462646,
(float)0.469757, (float)0.476868, (float)0.483978,
(float)0.491089, (float)0.498230, (float)0.505341,
(float)0.512451, (float)0.519592, (float)0.526703,
(float)0.533813, (float)0.540924, (float)0.548004,
(float)0.555084, (float)0.562164, (float)0.569244,
(float)0.576294, (float)0.583313, (float)0.590332,
(float)0.597321, (float)0.604309, (float)0.611267,
(float)0.618195, (float)0.625092, (float)0.631989,
(float)0.638855, (float)0.645660, (float)0.652466,
(float)0.659241, (float)0.665985, (float)0.672668,
(float)0.679352, (float)0.685974, (float)0.692566,
(float)0.699127, (float)0.705658, (float)0.712128,
(float)0.718536, (float)0.724945, (float)0.731262,
(float)0.737549, (float)0.743805, (float)0.750000,
(float)0.756134, (float)0.762238, (float)0.768280,
(float)0.774261, (float)0.780182, (float)0.786072,
(float)0.791870, (float)0.797638, (float)0.803314,
(float)0.808960, (float)0.814514, (float)0.820038,
(float)0.825470, (float)0.830841, (float)0.836151,
(float)0.841400, (float)0.846558, (float)0.851654,
(float)0.856689, (float)0.861633, (float)0.866516,
(float)0.871338, (float)0.876068, (float)0.880737,
(float)0.885315, (float)0.889801, (float)0.894226,
(float)0.898560, (float)0.902832, (float)0.907013,
(float)0.911102, (float)0.915100, (float)0.919037,
(float)0.922882, (float)0.926636, (float)0.930328,
(float)0.933899, (float)0.937408, (float)0.940796,
(float)0.944122, (float)0.947357, (float)0.950470,
(float)0.953522, (float)0.956482, (float)0.959351,
(float)0.962097, (float)0.964783, (float)0.967377,
(float)0.969849, (float)0.972229, (float)0.974518,
(float)0.976715, (float)0.978821, (float)0.980835,
(float)0.982727, (float)0.984528, (float)0.986237,
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(float)0.987854, (float)0.989380, (float)0.990784,
(float)0.992096, (float)0.993317, (float)0.994415,
(float)0.995422, (float)0.996338, (float)0.997162,
(float)0.997864, (float)0.998474, (float)0.998962,
(float)0.999390, (float)0.999695, (float)0.999878,
(float)0.999969, (float)0.999969, (float)0.996918,
(float)0.987701, (float)0.972382, (float)0.951050,
(float)0.923889, (float)0.891022, (float)0.852631,
(float)0.809021, (float)0.760406, (float)0.707092,
(float)0.649445, (float)0.587799, (float)0.522491,
(float)0.453979, (float)0.382690, (float)0.309021,
(float)0.233459, (float)0.156433, (float)0.078461
};
/* Lag window for LPC */
float lpc_lagwinTbl[LPC_FILTERORDER + 1]={
(float)1.000100, (float)0.998890, (float)0.995569,
(float)0.990057, (float)0.982392,
(float)0.972623, (float)0.960816, (float)0.947047,
(float)0.931405, (float)0.913989, (float)0.894909};
/* LSF quantization*/
float lsfCbTbl[64 * 3 + 128 * 3 + 128 * 4] = {
(float)0.155396, (float)0.273193, (float)0.451172,
(float)0.390503, (float)0.648071, (float)1.002075,
(float)0.440186, (float)0.692261, (float)0.955688,
(float)0.343628, (float)0.642334, (float)1.071533,
(float)0.318359, (float)0.491577, (float)0.670532,
(float)0.193115, (float)0.375488, (float)0.725708,
(float)0.364136, (float)0.510376, (float)0.658691,
(float)0.297485, (float)0.527588, (float)0.842529,
(float)0.227173, (float)0.365967, (float)0.563110,
(float)0.244995, (float)0.396729, (float)0.636475,
(float)0.169434, (float)0.300171, (float)0.520264,
(float)0.312866, (float)0.464478, (float)0.643188,
(float)0.248535, (float)0.429932, (float)0.626099,
(float)0.236206, (float)0.491333, (float)0.817139,
(float)0.334961, (float)0.625122, (float)0.895752,
(float)0.343018, (float)0.518555, (float)0.698608,
(float)0.372803, (float)0.659790, (float)0.945435,
(float)0.176880, (float)0.316528, (float)0.581421,
(float)0.416382, (float)0.625977, (float)0.805176,
(float)0.303223, (float)0.568726, (float)0.915039,
(float)0.203613, (float)0.351440, (float)0.588135,
(float)0.221191, (float)0.375000, (float)0.614746,
(float)0.199951, (float)0.323364, (float)0.476074,
(float)0.300781, (float)0.433350, (float)0.566895,
(float)0.226196, (float)0.354004, (float)0.507568,
(float)0.300049, (float)0.508179, (float)0.711670,
(float)0.312012, (float)0.492676, (float)0.763428,
(float)0.329956, (float)0.541016, (float)0.795776,
(float)0.373779, (float)0.604614, (float)0.928833,
(float)0.210571, (float)0.452026, (float)0.755249,
(float)0.271118, (float)0.473267, (float)0.662476,
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(float)0.285522, (float)0.436890, (float)0.634399,
(float)0.246704, (float)0.565552, (float)0.859009,
(float)0.270508, (float)0.406250, (float)0.553589,
(float)0.361450, (float)0.578491, (float)0.813843,
(float)0.342651, (float)0.482788, (float)0.622437,
(float)0.340332, (float)0.549438, (float)0.743164,
(float)0.200439, (float)0.336304, (float)0.540894,
(float)0.407837, (float)0.644775, (float)0.895142,
(float)0.294678, (float)0.454834, (float)0.699097,
(float)0.193115, (float)0.344482, (float)0.643188,
(float)0.275757, (float)0.420776, (float)0.598755,
(float)0.380493, (float)0.608643, (float)0.861084,
(float)0.222778, (float)0.426147, (float)0.676514,
(float)0.407471, (float)0.700195, (float)1.053101,
(float)0.218384, (float)0.377197, (float)0.669922,
(float)0.313232, (float)0.454102, (float)0.600952,
(float)0.347412, (float)0.571533, (float)0.874146,
(float)0.238037, (float)0.405396, (float)0.729492,
(float)0.223877, (float)0.412964, (float)0.822021,
(float)0.395264, (float)0.582153, (float)0.743896,
(float)0.247925, (float)0.485596, (float)0.720581,
(float)0.229126, (float)0.496582, (float)0.907715,
(float)0.260132, (float)0.566895, (float)1.012695,
(float)0.337402, (float)0.611572, (float)0.978149,
(float)0.267822, (float)0.447632, (float)0.769287,
(float)0.250610, (float)0.381714, (float)0.530029,
(float)0.430054, (float)0.805054, (float)1.221924,
(float)0.382568, (float)0.544067, (float)0.701660,
(float)0.383545, (float)0.710327, (float)1.149170,
(float)0.271362, (float)0.529053, (float)0.775513,
(float)0.246826, (float)0.393555, (float)0.588623,
(float)0.266846, (float)0.422119, (float)0.676758,
(float)0.311523, (float)0.580688, (float)0.838623,
(float)1.331177, (float)1.576782, (float)1.779541,
(float)1.160034, (float)1.401978, (float)1.768188,
(float)1.161865, (float)1.525146, (float)1.715332,
(float)0.759521, (float)0.913940, (float)1.119873,
(float)0.947144, (float)1.121338, (float)1.282471,
(float)1.015015, (float)1.557007, (float)1.804932,
(float)1.172974, (float)1.402100, (float)1.692627,
(float)1.087524, (float)1.474243, (float)1.665405,
(float)0.899536, (float)1.105225, (float)1.406250,
(float)1.148438, (float)1.484741, (float)1.796265,
(float)0.785645, (float)1.209839, (float)1.567749,
(float)0.867798, (float)1.166504, (float)1.450684,
(float)0.922485, (float)1.229858, (float)1.420898,
(float)0.791260, (float)1.123291, (float)1.409546,
(float)0.788940, (float)0.966064, (float)1.340332,
(float)1.051147, (float)1.272827, (float)1.556641,
(float)0.866821, (float)1.181152, (float)1.538818,
(float)0.906738, (float)1.373535, (float)1.607910,
(float)1.244751, (float)1.581421, (float)1.933838,
(float)0.913940, (float)1.337280, (float)1.539673,
(float)0.680542, (float)0.959229, (float)1.662720,
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(float)0.887207, (float)1.430542, (float)1.800781,
(float)0.912598, (float)1.433594, (float)1.683960,
(float)0.860474, (float)1.060303, (float)1.455322,
(float)1.005127, (float)1.381104, (float)1.706909,
(float)0.800781, (float)1.363892, (float)1.829102,
(float)0.781860, (float)1.124390, (float)1.505981,
(float)1.003662, (float)1.471436, (float)1.684692,
(float)0.981323, (float)1.309570, (float)1.618042,
(float)1.228760, (float)1.554321, (float)1.756470,
(float)0.734375, (float)0.895752, (float)1.225586,
(float)0.841797, (float)1.055664, (float)1.249268,
(float)0.920166, (float)1.119385, (float)1.486206,
(float)0.894409, (float)1.539063, (float)1.828979,
(float)1.283691, (float)1.543335, (float)1.858276,
(float)0.676025, (float)0.933105, (float)1.490845,
(float)0.821289, (float)1.491821, (float)1.739868,
(float)0.923218, (float)1.144653, (float)1.580566,
(float)1.057251, (float)1.345581, (float)1.635864,
(float)0.888672, (float)1.074951, (float)1.353149,
(float)0.942749, (float)1.195435, (float)1.505493,
(float)1.492310, (float)1.788086, (float)2.039673,
(float)1.070313, (float)1.634399, (float)1.860962,
(float)1.253296, (float)1.488892, (float)1.686035,
(float)0.647095, (float)0.864014, (float)1.401855,
(float)0.866699, (float)1.254883, (float)1.453369,
(float)1.063965, (float)1.532593, (float)1.731323,
(float)1.167847, (float)1.521484, (float)1.884033,
(float)0.956055, (float)1.502075, (float)1.745605,
(float)0.928711, (float)1.288574, (float)1.479614,
(float)1.088013, (float)1.380737, (float)1.570801,
(float)0.905029, (float)1.186768, (float)1.371948,
(float)1.057861, (float)1.421021, (float)1.617432,
(float)1.108276, (float)1.312500, (float)1.501465,
(float)0.979492, (float)1.416992, (float)1.624268,
(float)1.276001, (float)1.661011, (float)2.007935,
(float)0.993042, (float)1.168579, (float)1.331665,
(float)0.778198, (float)0.944946, (float)1.235962,
(float)1.223755, (float)1.491333, (float)1.815674,
(float)0.852661, (float)1.350464, (float)1.722290,
(float)1.134766, (float)1.593140, (float)1.787354,
(float)1.051392, (float)1.339722, (float)1.531006,
(float)0.803589, (float)1.271240, (float)1.652100,
(float)0.755737, (float)1.143555, (float)1.639404,
(float)0.700928, (float)0.837280, (float)1.130371,
(float)0.942749, (float)1.197876, (float)1.669800,
(float)0.993286, (float)1.378296, (float)1.566528,
(float)0.801025, (float)1.095337, (float)1.298950,
(float)0.739990, (float)1.032959, (float)1.383667,
(float)0.845703, (float)1.072266, (float)1.543823,
(float)0.915649, (float)1.072266, (float)1.224487,
(float)1.021973, (float)1.226196, (float)1.481323,
(float)0.999878, (float)1.204102, (float)1.555908,
(float)0.722290, (float)0.913940, (float)1.340210,
(float)0.673340, (float)0.835938, (float)1.259521,
Andersen et. al. Experimental - Expires April 25th, 2004 85
Internet Low Bit Rate Codec October 2003
(float)0.832397, (float)1.208374, (float)1.394165,
(float)0.962158, (float)1.576172, (float)1.912842,
(float)1.166748, (float)1.370850, (float)1.556763,
(float)0.946289, (float)1.138550, (float)1.400391,
(float)1.035034, (float)1.218262, (float)1.386475,
(float)1.393799, (float)1.717773, (float)2.000244,
(float)0.972656, (float)1.260986, (float)1.760620,
(float)1.028198, (float)1.288452, (float)1.484619,
(float)0.773560, (float)1.258057, (float)1.756714,
(float)1.080322, (float)1.328003, (float)1.742676,
(float)0.823975, (float)1.450806, (float)1.917725,
(float)0.859009, (float)1.016602, (float)1.191895,
(float)0.843994, (float)1.131104, (float)1.645020,
(float)1.189697, (float)1.702759, (float)1.894409,
(float)1.346680, (float)1.763184, (float)2.066040,
(float)0.980469, (float)1.253784, (float)1.441650,
(float)1.338135, (float)1.641968, (float)1.932739,
(float)1.223267, (float)1.424194, (float)1.626465,
(float)0.765747, (float)1.004150, (float)1.579102,
(float)1.042847, (float)1.269165, (float)1.647461,
(float)0.968750, (float)1.257568, (float)1.555786,
(float)0.826294, (float)0.993408, (float)1.275146,
(float)0.742310, (float)0.950439, (float)1.430542,
(float)1.054321, (float)1.439819, (float)1.828003,
(float)1.072998, (float)1.261719, (float)1.441895,
(float)0.859375, (float)1.036377, (float)1.314819,
(float)0.895752, (float)1.267212, (float)1.605591,
(float)0.805420, (float)0.962891, (float)1.142334,
(float)0.795654, (float)1.005493, (float)1.468506,
(float)1.105347, (float)1.313843, (float)1.584839,
(float)0.792236, (float)1.221802, (float)1.465698,
(float)1.170532, (float)1.467651, (float)1.664063,
(float)0.838257, (float)1.153198, (float)1.342163,
(float)0.968018, (float)1.198242, (float)1.391235,
(float)1.250122, (float)1.623535, (float)1.823608,
(float)0.711670, (float)1.058350, (float)1.512085,
(float)1.204834, (float)1.454468, (float)1.739136,
(float)1.137451, (float)1.421753, (float)1.620117,
(float)0.820435, (float)1.322754, (float)1.578247,
(float)0.798706, (float)1.005005, (float)1.213867,
(float)0.980713, (float)1.324951, (float)1.512939,
(float)1.112305, (float)1.438843, (float)1.735596,
(float)1.135498, (float)1.356689, (float)1.635742,
(float)1.101318, (float)1.387451, (float)1.686523,
(float)0.849854, (float)1.276978, (float)1.523438,
(float)1.377930, (float)1.627563, (float)1.858154,
(float)0.884888, (float)1.095459, (float)1.287476,
(float)1.289795, (float)1.505859, (float)1.756592,
(float)0.817505, (float)1.384155, (float)1.650513,
(float)1.446655, (float)1.702148, (float)1.931885,
(float)0.835815, (float)1.023071, (float)1.385376,
(float)0.916626, (float)1.139038, (float)1.335327,
(float)0.980103, (float)1.174072, (float)1.453735,
(float)1.705688, (float)2.153809, (float)2.398315, (float)2.743408,
Andersen et. al. Experimental - Expires April 25th, 2004 86
Internet Low Bit Rate Codec October 2003
(float)1.797119, (float)2.016846, (float)2.445679, (float)2.701904,
(float)1.990356, (float)2.219116, (float)2.576416, (float)2.813477,
(float)1.849365, (float)2.190918, (float)2.611572, (float)2.835083,
(float)1.657959, (float)1.854370, (float)2.159058, (float)2.726196,
(float)1.437744, (float)1.897705, (float)2.253174, (float)2.655396,
(float)2.028687, (float)2.247314, (float)2.542358, (float)2.875854,
(float)1.736938, (float)1.922119, (float)2.185913, (float)2.743408,
(float)1.521606, (float)1.870972, (float)2.526855, (float)2.786987,
(float)1.841431, (float)2.050659, (float)2.463623, (float)2.857666,
(float)1.590088, (float)2.067261, (float)2.427979, (float)2.794434,
(float)1.746826, (float)2.057373, (float)2.320190, (float)2.800781,
(float)1.734619, (float)1.940552, (float)2.306030, (float)2.826416,
(float)1.786255, (float)2.204468, (float)2.457520, (float)2.795288,
(float)1.861084, (float)2.170532, (float)2.414551, (float)2.763672,
(float)2.001465, (float)2.307617, (float)2.552734, (float)2.811890,
(float)1.784424, (float)2.124146, (float)2.381592, (float)2.645508,
(float)1.888794, (float)2.135864, (float)2.418579, (float)2.861206,
(float)2.301147, (float)2.531250, (float)2.724976, (float)2.913086,
(float)1.837769, (float)2.051270, (float)2.261963, (float)2.553223,
(float)2.012939, (float)2.221191, (float)2.440186, (float)2.678101,
(float)1.429565, (float)1.858276, (float)2.582275, (float)2.845703,
(float)1.622803, (float)1.897705, (float)2.367310, (float)2.621094,
(float)1.581543, (float)1.960449, (float)2.515869, (float)2.736450,
(float)1.419434, (float)1.933960, (float)2.394653, (float)2.746704,
(float)1.721924, (float)2.059570, (float)2.421753, (float)2.769653,
(float)1.911011, (float)2.220703, (float)2.461060, (float)2.740723,
(float)1.581177, (float)1.860840, (float)2.516968, (float)2.874634,
(float)1.870361, (float)2.098755, (float)2.432373, (float)2.656494,
(float)2.059692, (float)2.279785, (float)2.495605, (float)2.729370,
(float)1.815674, (float)2.181519, (float)2.451538, (float)2.680542,
(float)1.407959, (float)1.768311, (float)2.343018, (float)2.668091,
(float)2.168701, (float)2.394653, (float)2.604736, (float)2.829346,
(float)1.636230, (float)1.865723, (float)2.329102, (float)2.824219,
(float)1.878906, (float)2.139526, (float)2.376709, (float)2.679810,
(float)1.765381, (float)1.971802, (float)2.195435, (float)2.586914,
(float)2.164795, (float)2.410889, (float)2.673706, (float)2.903198,
(float)2.071899, (float)2.331055, (float)2.645874, (float)2.907104,
(float)2.026001, (float)2.311523, (float)2.594849, (float)2.863892,
(float)1.948975, (float)2.180786, (float)2.514893, (float)2.797852,
(float)1.881836, (float)2.130859, (float)2.478149, (float)2.804199,
(float)2.238159, (float)2.452759, (float)2.652832, (float)2.868286,
(float)1.897949, (float)2.101685, (float)2.524292, (float)2.880127,
(float)1.856445, (float)2.074585, (float)2.541016, (float)2.791748,
(float)1.695557, (float)2.199097, (float)2.506226, (float)2.742676,
(float)1.612671, (float)1.877075, (float)2.435425, (float)2.732910,
(float)1.568848, (float)1.786499, (float)2.194580, (float)2.768555,
(float)1.953369, (float)2.164551, (float)2.486938, (float)2.874023,
(float)1.388306, (float)1.725342, (float)2.384521, (float)2.771851,
(float)2.115356, (float)2.337769, (float)2.592896, (float)2.864014,
(float)1.905762, (float)2.111328, (float)2.363525, (float)2.789307,
(float)1.882568, (float)2.332031, (float)2.598267, (float)2.827637,
(float)1.683594, (float)2.088745, (float)2.361938, (float)2.608643,
(float)1.874023, (float)2.182129, (float)2.536133, (float)2.766968,
Andersen et. al. Experimental - Expires April 25th, 2004 87
Internet Low Bit Rate Codec October 2003
(float)1.861938, (float)2.070435, (float)2.309692, (float)2.700562,
(float)1.722168, (float)2.107422, (float)2.477295, (float)2.837646,
(float)1.926880, (float)2.184692, (float)2.442627, (float)2.663818,
(float)2.123901, (float)2.337280, (float)2.553101, (float)2.777466,
(float)1.588135, (float)1.911499, (float)2.212769, (float)2.543945,
(float)2.053955, (float)2.370850, (float)2.712158, (float)2.939941,
(float)2.210449, (float)2.519653, (float)2.770386, (float)2.958618,
(float)2.199463, (float)2.474731, (float)2.718262, (float)2.919922,
(float)1.960083, (float)2.175415, (float)2.608032, (float)2.888794,
(float)1.953735, (float)2.185181, (float)2.428223, (float)2.809570,
(float)1.615234, (float)2.036499, (float)2.576538, (float)2.834595,
(float)1.621094, (float)2.028198, (float)2.431030, (float)2.664673,
(float)1.824951, (float)2.267456, (float)2.514526, (float)2.747925,
(float)1.994263, (float)2.229126, (float)2.475220, (float)2.833984,
(float)1.746338, (float)2.011353, (float)2.588257, (float)2.826904,
(float)1.562866, (float)2.135986, (float)2.471680, (float)2.687256,
(float)1.748901, (float)2.083496, (float)2.460938, (float)2.686279,
(float)1.758057, (float)2.131470, (float)2.636597, (float)2.891602,
(float)2.071289, (float)2.299072, (float)2.550781, (float)2.814331,
(float)1.839600, (float)2.094360, (float)2.496460, (float)2.723999,
(float)1.882202, (float)2.088257, (float)2.636841, (float)2.923096,
(float)1.957886, (float)2.153198, (float)2.384399, (float)2.615234,
(float)1.992920, (float)2.351196, (float)2.654419, (float)2.889771,
(float)2.012817, (float)2.262451, (float)2.643799, (float)2.903076,
(float)2.025635, (float)2.254761, (float)2.508423, (float)2.784058,
(float)2.316040, (float)2.589355, (float)2.794189, (float)2.963623,
(float)1.741211, (float)2.279541, (float)2.578491, (float)2.816284,
(float)1.845337, (float)2.055786, (float)2.348511, (float)2.822021,
(float)1.679932, (float)1.926514, (float)2.499756, (float)2.835693,
(float)1.722534, (float)1.946899, (float)2.448486, (float)2.728760,
(float)1.829834, (float)2.043213, (float)2.580444, (float)2.867676,
(float)1.676636, (float)2.071655, (float)2.322510, (float)2.704834,
(float)1.791504, (float)2.113525, (float)2.469727, (float)2.784058,
(float)1.977051, (float)2.215088, (float)2.497437, (float)2.726929,
(float)1.800171, (float)2.106689, (float)2.357788, (float)2.738892,
(float)1.827759, (float)2.170166, (float)2.525879, (float)2.852417,
(float)1.918335, (float)2.132813, (float)2.488403, (float)2.728149,
(float)1.916748, (float)2.225098, (float)2.542603, (float)2.857666,
(float)1.761230, (float)1.976074, (float)2.507446, (float)2.884521,
(float)2.053711, (float)2.367432, (float)2.608032, (float)2.837646,
(float)1.595337, (float)2.000977, (float)2.307129, (float)2.578247,
(float)1.470581, (float)2.031250, (float)2.375854, (float)2.647583,
(float)1.801392, (float)2.128052, (float)2.399780, (float)2.822876,
(float)1.853638, (float)2.066650, (float)2.429199, (float)2.751465,
(float)1.956299, (float)2.163696, (float)2.394775, (float)2.734253,
(float)1.963623, (float)2.275757, (float)2.585327, (float)2.865234,
(float)1.887451, (float)2.105469, (float)2.331787, (float)2.587402,
(float)2.120117, (float)2.443359, (float)2.733887, (float)2.941406,
(float)1.506348, (float)1.766968, (float)2.400513, (float)2.851807,
(float)1.664551, (float)1.981079, (float)2.375732, (float)2.774414,
(float)1.720703, (float)1.978882, (float)2.391479, (float)2.640991,
(float)1.483398, (float)1.814819, (float)2.434448, (float)2.722290,
(float)1.769043, (float)2.136597, (float)2.563721, (float)2.774414,
(float)1.810791, (float)2.049316, (float)2.373901, (float)2.613647,
Andersen et. al. Experimental - Expires April 25th, 2004 88
Internet Low Bit Rate Codec October 2003
(float)1.788330, (float)2.005981, (float)2.359131, (float)2.723145,
(float)1.785156, (float)1.993164, (float)2.399780, (float)2.832520,
(float)1.695313, (float)2.022949, (float)2.522583, (float)2.745117,
(float)1.584106, (float)1.965576, (float)2.299927, (float)2.715576,
(float)1.894897, (float)2.249878, (float)2.655884, (float)2.897705,
(float)1.720581, (float)1.995728, (float)2.299438, (float)2.557007,
(float)1.619385, (float)2.173950, (float)2.574219, (float)2.787964,
(float)1.883179, (float)2.220459, (float)2.474365, (float)2.825073,
(float)1.447632, (float)2.045044, (float)2.555542, (float)2.744873,
(float)1.502686, (float)2.156616, (float)2.653320, (float)2.846558,
(float)1.711548, (float)1.944092, (float)2.282959, (float)2.685791,
(float)1.499756, (float)1.867554, (float)2.341064, (float)2.578857,
(float)1.916870, (float)2.135132, (float)2.568237, (float)2.826050,
(float)1.498047, (float)1.711182, (float)2.223267, (float)2.755127,
(float)1.808716, (float)1.997559, (float)2.256470, (float)2.758545,
(float)2.088501, (float)2.402710, (float)2.667358, (float)2.890259,
(float)1.545044, (float)1.819214, (float)2.324097, (float)2.692993,
(float)1.796021, (float)2.012573, (float)2.505737, (float)2.784912,
(float)1.786499, (float)2.041748, (float)2.290405, (float)2.650757,
(float)1.938232, (float)2.264404, (float)2.529053, (float)2.796143
};
A.9 anaFilter.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
anaFilter.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_ANAFILTER_H
#define __iLBC_ANAFILTER_H
void anaFilter(
float *In, /* (i) Signal to be filtered */
float *a, /* (i) LP parameters */
int len,/* (i) Length of signal */
float *Out, /* (o) Filtered signal */
float *mem /* (i/o) Filter state */
);
#endif
A.10 anaFilter.c
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/******************************************************************
iLBC Speech Coder ANSI-C Source Code
anaFilter.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <string.h>
#include "iLBC_define.h"
/*----------------------------------------------------------------*
* LP analysis filter.
*---------------------------------------------------------------*/
void anaFilter(
float *In, /* (i) Signal to be filtered */
float *a, /* (i) LP parameters */
int len,/* (i) Length of signal */
float *Out, /* (o) Filtered signal */
float *mem /* (i/o) Filter state */
){
int i, j;
float *po, *pi, *pm, *pa;
po = Out;
/* Filter first part using memory from past */
for (i=0; i<LPC_FILTERORDER; i++) {
pi = &In[i];
pm = &mem[LPC_FILTERORDER-1];
pa = a;
*po=0.0;
for (j=0; j<=i; j++) {
*po+=(*pa++)*(*pi--);
}
for (j=i+1; j<LPC_FILTERORDER+1; j++) {
*po+=(*pa++)*(*pm--);
}
po++;
}
/* Filter last part where the state is entierly
in the input vector */
for (i=LPC_FILTERORDER; i<len; i++) {
pi = &In[i];
pa = a;
*po=0.0;
Andersen et. al. Experimental - Expires April 25th, 2004 90
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for (j=0; j<LPC_FILTERORDER+1; j++) {
*po+=(*pa++)*(*pi--);
}
po++;
}
/* Update state vector */
memcpy(mem, &In[len-LPC_FILTERORDER],
LPC_FILTERORDER*sizeof(float));
}
A.11 createCB.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
createCB.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_CREATECB_H
#define __iLBC_CREATECB_H
void filteredCBvecs(
float *cbvectors, /* (o) Codebook vector for the
higher section */
float *mem, /* (i) Buffer to create codebook
vectors from */
int lMem /* (i) Length of buffer */
);
void searchAugmentedCB(
int low, /* (i) Start index for the search */
int high, /* (i) End index for the search */
int stage, /* (i) Current stage */
int startIndex, /* (i) CB index for the first
augmented vector */
float *target, /* (i) Target vector for encoding */
float *buffer, /* (i) Pointer to the end of the
buffer for augmented codebook
construction */
float *max_measure, /* (i/o) Currently maximum measure */
int *best_index,/* (o) Currently the best index */
float *gain, /* (o) Currently the best gain */
float *energy, /* (o) Energy of augmented
codebook vectors */
float *invenergy/* (o) Inv energy of aug codebook
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vectors */
);
void createAugmentedVec(
int index, /* (i) Index for the aug vector
to be created */
float *buffer, /* (i) Pointer to the end of the
buffer for augmented codebook
construction */
float *cbVec /* (o) The construced codebook vector */
);
#endif
A.12 createCB.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
createCB.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "iLBC_define.h"
#include "constants.h"
#include <string.h>
#include <math.h>
/*----------------------------------------------------------------*
* Construct an additional codebook vector by filtering the
* initial codebook buffer. This vector is then used to expand
* the codebook with an additional section.
*---------------------------------------------------------------*/
void filteredCBvecs(
float *cbvectors, /* (o) Codebook vectors for the
higher section */
float *mem, /* (i) Buffer to create codebook
vector from */
int lMem /* (i) Length of buffer */
){
int j, k;
float *pp, *pp1;
float tempbuff2[CB_MEML+CB_FILTERLEN];
float *pos;
memset(tempbuff2, 0, (CB_HALFFILTERLEN-1)*sizeof(float));
memcpy(&tempbuff2[CB_HALFFILTERLEN-1], mem, lMem*sizeof(float));
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memset(&tempbuff2[lMem+CB_HALFFILTERLEN-1], 0,
(CB_HALFFILTERLEN+1)*sizeof(float));
/* Create codebook vector for higher section by filtering */
/* do filtering */
pos=cbvectors;
memset(pos, 0, lMem*sizeof(float));
for (k=0; k<lMem; k++) {
pp=&tempbuff2[k];
pp1=&cbfiltersTbl[CB_FILTERLEN-1];
for (j=0;j<CB_FILTERLEN;j++) {
(*pos)+=(*pp++)*(*pp1--);
}
pos++;
}
}
/*----------------------------------------------------------------*
* Search the augmented part of the codebook to find the best
* measure.
*----------------------------------------------------------------*/
void searchAugmentedCB(
int low, /* (i) Start index for the search */
int high, /* (i) End index for the search */
int stage, /* (i) Current stage */
int startIndex, /* (i) Codebook index for the first
aug vector */
float *target, /* (i) Target vector for encoding */
float *buffer, /* (i) Pointer to the end of the buffer for
augmented codebook construction */
float *max_measure, /* (i/o) Currently maximum measure */
int *best_index,/* (o) Currently the best index */
float *gain, /* (o) Currently the best gain */
float *energy, /* (o) Energy of augmented codebook
vectors */
float *invenergy/* (o) Inv energy of augmented codebook
vectors */
) {
int icount, ilow, j, tmpIndex;
float *pp, *ppo, *ppi, *ppe, crossDot, alfa;
float weighted, measure, nrjRecursive;
float ftmp;
/* Compute the energy for the first (low-5)
noninterpolated samples */
nrjRecursive = (float) 0.0;
pp = buffer - low + 1;
for (j=0; j<(low-5); j++) {
nrjRecursive += ( (*pp)*(*pp) );
pp++;
}
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ppe = buffer - low;
for (icount=low; icount<=high; icount++) {
/* Index of the codebook vector used for retrieving
energy values */
tmpIndex = startIndex+icount-20;
ilow = icount-4;
/* Update the energy recursively to save complexity */
nrjRecursive = nrjRecursive + (*ppe)*(*ppe);
ppe--;
energy[tmpIndex] = nrjRecursive;
/* Compute cross dot product for the first (low-5)
samples */
crossDot = (float) 0.0;
pp = buffer-icount;
for (j=0; j<ilow; j++) {
crossDot += target[j]*(*pp++);
}
/* interpolation */
alfa = (float) 0.2;
ppo = buffer-4;
ppi = buffer-icount-4;
for (j=ilow; j<icount; j++) {
weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
ppo++;
ppi++;
energy[tmpIndex] += weighted*weighted;
crossDot += target[j]*weighted;
alfa += (float)0.2;
}
/* Compute energy and cross dot product for the
remaining samples */
pp = buffer - icount;
for (j=icount; j<SUBL; j++) {
energy[tmpIndex] += (*pp)*(*pp);
crossDot += target[j]*(*pp++);
}
if (energy[tmpIndex]>0.0) {
invenergy[tmpIndex]=(float)1.0/(energy[tmpIndex]+EPS);
} else {
invenergy[tmpIndex] = (float) 0.0;
}
if (stage==0) {
measure = (float)-10000000.0;
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if (crossDot > 0.0) {
measure = crossDot*crossDot*invenergy[tmpIndex];
}
}
else {
measure = crossDot*crossDot*invenergy[tmpIndex];
}
/* check if measure is better */
ftmp = crossDot*invenergy[tmpIndex];
if ((measure>*max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
*best_index = tmpIndex;
*max_measure = measure;
*gain = ftmp;
}
}
}
/*----------------------------------------------------------------*
* Recreate a specific codebook vector from the augmented part.
*
*----------------------------------------------------------------*/
void createAugmentedVec(
int index, /* (i) Index for the augmented vector
to be created */
float *buffer, /* (i) Pointer to the end of the buffer for
augmented codebook construction */
float *cbVec/* (o) The construced codebook vector */
) {
int ilow, j;
float *pp, *ppo, *ppi, alfa, alfa1, weighted;
ilow = index-5;
/* copy the first noninterpolated part */
pp = buffer-index;
memcpy(cbVec,pp,sizeof(float)*index);
/* interpolation */
alfa1 = (float)0.2;
alfa = 0.0;
ppo = buffer-5;
ppi = buffer-index-5;
for (j=ilow; j<index; j++) {
weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
ppo++;
ppi++;
cbVec[j] = weighted;
alfa += alfa1;
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}
/* copy the second noninterpolated part */
pp = buffer - index;
memcpy(cbVec+index,pp,sizeof(float)*(SUBL-index));
}
A.13 doCPLC.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
doCPLC.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_DOLPC_H
#define __iLBC_DOLPC_H
void doThePLC(
float *PLCresidual, /* (o) concealed residual */
float *PLClpc, /* (o) concealed LP parameters */
int PLI, /* (i) packet loss indicator
0 - no PL, 1 = PL */
float *decresidual, /* (i) decoded residual */
float *lpc, /* (i) decoded LPC (only used for no PL) */
int inlag, /* (i) pitch lag */
iLBC_Dec_Inst_t *iLBCdec_inst
/* (i/o) decoder instance */
);
#endif
A.14 doCPLC.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
doCPLC.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
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#include <math.h>
#include <string.h>
#include <stdio.h>
#include "iLBC_define.h"
/*----------------------------------------------------------------*
* Compute cross correlation and pitch gain for pitch prediction
* of last subframe at given lag.
*---------------------------------------------------------------*/
void compCorr(
float *cc, /* (o) cross correlation coefficient */
float *gc, /* (o) gain */
float *pm,
float *buffer, /* (i) signal buffer */
int lag, /* (i) pitch lag */
int bLen, /* (i) length of buffer */
int sRange /* (i) correlation search length */
){
int i;
float ftmp1, ftmp2, ftmp3;
/* Guard against getting outside buffer */
if ((bLen-sRange-lag)<0) {
sRange=bLen-lag;
}
ftmp1 = 0.0;
ftmp2 = 0.0;
ftmp3 = 0.0;
for (i=0; i<sRange; i++) {
ftmp1 += buffer[bLen-sRange+i] *
buffer[bLen-sRange+i-lag];
ftmp2 += buffer[bLen-sRange+i-lag] *
buffer[bLen-sRange+i-lag];
ftmp3 += buffer[bLen-sRange+i] *
buffer[bLen-sRange+i];
}
if (ftmp2 > 0.0) {
*cc = ftmp1*ftmp1/ftmp2;
*gc = (float)fabs(ftmp1/ftmp2);
*pm=(float)fabs(ftmp1)/
((float)sqrt(ftmp2)*(float)sqrt(ftmp3));
}
else {
*cc = 0.0;
*gc = 0.0;
*pm=0.0;
}
}
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/*----------------------------------------------------------------*
* Packet loss concealment routine. Conceals a residual signal
* and LP parameters. If no packet loss, update state.
*---------------------------------------------------------------*/
void doThePLC(
float *PLCresidual, /* (o) concealed residual */
float *PLClpc, /* (o) concealed LP parameters */
int PLI, /* (i) packet loss indicator
0 - no PL, 1 = PL */
float *decresidual, /* (i) decoded residual */
float *lpc, /* (i) decoded LPC (only used for no PL) */
int inlag, /* (i) pitch lag */
iLBC_Dec_Inst_t *iLBCdec_inst
/* (i/o) decoder instance */
){
int lag=20, randlag;
float gain, maxcc;
float use_gain;
float gain_comp, maxcc_comp, per, max_per;
int i, pick, use_lag;
float ftmp, randvec[BLOCKL_MAX], pitchfact, energy;
/* Packet Loss */
if (PLI == 1) {
iLBCdec_inst->consPLICount += 1;
/* if previous frame not lost,
determine pitch pred. gain */
if (iLBCdec_inst->prevPLI != 1) {
/* Search around the previous lag to find the
best pitch period */
lag=inlag-3;
compCorr(&maxcc, &gain, &max_per,
iLBCdec_inst->prevResidual,
lag, iLBCdec_inst->blockl, 60);
for (i=inlag-2;i<=inlag+3;i++) {
compCorr(&maxcc_comp, &gain_comp, &per,
iLBCdec_inst->prevResidual,
i, iLBCdec_inst->blockl, 60);
if (maxcc_comp>maxcc) {
maxcc=maxcc_comp;
gain=gain_comp;
lag=i;
max_per=per;
}
}
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}
/* previous frame lost, use recorded lag and periodicity */
else {
lag=iLBCdec_inst->prevLag;
max_per=iLBCdec_inst->per;
}
/* downscaling */
use_gain=1.0;
if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320)
use_gain=(float)0.9;
else if (iLBCdec_inst->consPLICount*
iLBCdec_inst->blockl>2*320)
use_gain=(float)0.7;
else if (iLBCdec_inst->consPLICount*
iLBCdec_inst->blockl>3*320)
use_gain=(float)0.5;
else if (iLBCdec_inst->consPLICount*
iLBCdec_inst->blockl>4*320)
use_gain=(float)0.0;
/* mix noise and pitch repeatition */
ftmp=(float)sqrt(max_per);
if (ftmp>(float)0.7)
pitchfact=(float)1.0;
else if (ftmp>(float)0.4)
pitchfact=(ftmp-(float)0.4)/((float)0.7-(float)0.4);
else
pitchfact=0.0;
/* avoid repetition of same pitch cycle */
use_lag=lag;
if (lag<80) {
use_lag=2*lag;
}
/* compute concealed residual */
energy = 0.0;
for (i=0; i<iLBCdec_inst->blockl; i++) {
/* noise component */
iLBCdec_inst->seed=(iLBCdec_inst->seed*69069L+1) &
(0x80000000L-1);
randlag = 50 + ((signed long) iLBCdec_inst->seed)%70;
pick = i - randlag;
if (pick < 0) {
randvec[i] =
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iLBCdec_inst->prevResidual[
iLBCdec_inst->blockl+pick];
} else {
randvec[i] = randvec[pick];
}
/* pitch repeatition component */
pick = i - use_lag;
if (pick < 0) {
PLCresidual[i] =
iLBCdec_inst->prevResidual[
iLBCdec_inst->blockl+pick];
} else {
PLCresidual[i] = PLCresidual[pick];
}
/* mix random and periodicity component */
if (i<80)
PLCresidual[i] = use_gain*(pitchfact *
PLCresidual[i] +
((float)1.0 - pitchfact) * randvec[i]);
else if (i<160)
PLCresidual[i] = (float)0.95*use_gain*(pitchfact *
PLCresidual[i] +
((float)1.0 - pitchfact) * randvec[i]);
else
PLCresidual[i] = (float)0.9*use_gain*(pitchfact *
PLCresidual[i] +
((float)1.0 - pitchfact) * randvec[i]);
energy += PLCresidual[i] * PLCresidual[i];
}
/* less than 30 dB, use only noise */
if (sqrt(energy/(float)iLBCdec_inst->blockl) < 30.0) {
gain=0.0;
for (i=0; i<iLBCdec_inst->blockl; i++) {
PLCresidual[i] = randvec[i];
}
}
/* use old LPC */
memcpy(PLClpc,iLBCdec_inst->prevLpc,
(LPC_FILTERORDER+1)*sizeof(float));
}
/* no packet loss, copy input */
else {
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memcpy(PLCresidual, decresidual,
iLBCdec_inst->blockl*sizeof(float));
memcpy(PLClpc, lpc, (LPC_FILTERORDER+1)*sizeof(float));
iLBCdec_inst->consPLICount = 0;
}
/* update state */
if (PLI) {
iLBCdec_inst->prevLag = lag;
iLBCdec_inst->per=max_per;
}
iLBCdec_inst->prevPLI = PLI;
memcpy(iLBCdec_inst->prevLpc, PLClpc,
(LPC_FILTERORDER+1)*sizeof(float));
memcpy(iLBCdec_inst->prevResidual, PLCresidual,
iLBCdec_inst->blockl*sizeof(float));
}
A.15 enhancer.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
enhancer.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __ENHANCER_H
#define __ENHANCER_H
#include "iLBC_define.h"
float xCorrCoef(
float *target, /* (i) first array */
float *regressor, /* (i) second array */
int subl /* (i) dimension arrays */
);
int enhancerInterface(
float *out, /* (o) the enhanced recidual signal */
float *in, /* (i) the recidual signal to enhance */
iLBC_Dec_Inst_t *iLBCdec_inst
/* (i/o) the decoder state structure */
);
#endif
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A.16 enhancer.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
enhancer.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <string.h>
#include "iLBC_define.h"
#include "constants.h"
#include "filter.h"
/*----------------------------------------------------------------*
* Find index in array such that the array element with said
* index is the element of said array closest to "value"
* according to the squared-error criterion
*---------------------------------------------------------------*/
void NearestNeighbor(
int *index, /* (o) index of array element closest
to value */
float *array, /* (i) data array */
float value,/* (i) value */
int arlength/* (i) dimension of data array */
){
int i;
float bestcrit,crit;
crit=array[0]-value;
bestcrit=crit*crit;
*index=0;
for (i=1; i<arlength; i++) {
crit=array[i]-value;
crit=crit*crit;
if (crit<bestcrit) {
bestcrit=crit;
*index=i;
}
}
}
/*----------------------------------------------------------------*
* compute cross correlation between sequences
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*---------------------------------------------------------------*/
void mycorr1(
float* corr, /* (o) correlation of seq1 and seq2 */
float* seq1, /* (i) first sequence */
int dim1, /* (i) dimension first seq1 */
const float *seq2, /* (i) second sequence */
int dim2 /* (i) dimension seq2 */
){
int i,j;
for (i=0; i<=dim1-dim2; i++) {
corr[i]=0.0;
for (j=0; j<dim2; j++) {
corr[i] += seq1[i+j] * seq2[j];
}
}
}
/*----------------------------------------------------------------*
* upsample finite array assuming zeros outside bounds
*---------------------------------------------------------------*/
void enh_upsample(
float* useq1, /* (o) upsampled output sequence */
float* seq1,/* (i) unupsampled sequence */
int dim1, /* (i) dimension seq1 */
int hfl /* (i) polyphase filter length=2*hfl+1 */
){
float *pu,*ps;
int i,j,k,q,filterlength,hfl2;
const float *polyp[ENH_UPS0]; /* pointers to
polyphase columns */
const float *pp;
/* define pointers for filter */
filterlength=2*hfl+1;
if ( filterlength > dim1 ) {
hfl2=(int) (dim1/2);
for (j=0; j<ENH_UPS0; j++) {
polyp[j]=polyphaserTbl+j*filterlength+hfl-hfl2;
}
hfl=hfl2;
filterlength=2*hfl+1;
}
else {
for (j=0; j<ENH_UPS0; j++) {
polyp[j]=polyphaserTbl+j*filterlength;
}
}
/* filtering: filter overhangs left side of sequence */
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pu=useq1;
for (i=hfl; i<filterlength; i++) {
for (j=0; j<ENH_UPS0; j++) {
*pu=0.0;
pp = polyp[j];
ps = seq1+i;
for (k=0; k<=i; k++) {
*pu += *ps-- * *pp++;
}
pu++;
}
}
/* filtering: simple convolution=inner products */
for (i=filterlength; i<dim1; i++) {
for (j=0;j<ENH_UPS0; j++){
*pu=0.0;
pp = polyp[j];
ps = seq1+i;
for (k=0; k<filterlength; k++) {
*pu += *ps-- * *pp++;
}
pu++;
}
}
/* filtering: filter overhangs right side of sequence */
for (q=1; q<=hfl; q++) {
for (j=0; j<ENH_UPS0; j++) {
*pu=0.0;
pp = polyp[j]+q;
ps = seq1+dim1-1;
for (k=0; k<filterlength-q; k++) {
*pu += *ps-- * *pp++;
}
pu++;
}
}
}
/*----------------------------------------------------------------*
* find segment starting near idata+estSegPos that has highest
* correlation with idata+centerStartPos through
* idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
* resolution of ENH_UPSO times the original of the original
* sampling rate
*---------------------------------------------------------------*/
void refiner(
float *seg, /* (o) segment array */
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float *updStartPos, /* (o) updated start point */
float* idata, /* (i) original data buffer */
int idatal, /* (i) dimension of idata */
int centerStartPos, /* (i) beginning center segment */
float estSegPos,/* (i) estimated beginning other segment */
float period /* (i) estimated pitch period */
){
int estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
int tloc,tloc2,i,st,en,fraction;
float vect[ENH_VECTL],corrVec[ENH_CORRDIM],maxv;
float corrVecUps[ENH_CORRDIM*ENH_UPS0];
/* defining array bounds */
estSegPosRounded=(int)(estSegPos - 0.5);
searchSegStartPos=estSegPosRounded-ENH_SLOP;
if (searchSegStartPos<0) {
searchSegStartPos=0;
}
searchSegEndPos=estSegPosRounded+ENH_SLOP;
if (searchSegEndPos+ENH_BLOCKL >= idatal) {
searchSegEndPos=idatal-ENH_BLOCKL-1;
}
corrdim=searchSegEndPos-searchSegStartPos+1;
/* compute upsampled correlation (corr33) and find
location of max */
mycorr1(corrVec,idata+searchSegStartPos,
corrdim+ENH_BLOCKL-1,idata+centerStartPos,ENH_BLOCKL);
enh_upsample(corrVecUps,corrVec,corrdim,ENH_FL0);
tloc=0; maxv=corrVecUps[0];
for (i=1; i<ENH_UPS0*corrdim; i++) {
if (corrVecUps[i]>maxv) {
tloc=i;
maxv=corrVecUps[i];
}
}
/* make vector can be upsampled without ever running outside
bounds */
*updStartPos= (float)searchSegStartPos +
(float)tloc/(float)ENH_UPS0+(float)1.0;
tloc2=(int)(tloc/ENH_UPS0);
if (tloc>tloc2*ENH_UPS0) {
tloc2++;
}
st=searchSegStartPos+tloc2-ENH_FL0;
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if (st<0) {
memset(vect,0,-st*sizeof(float));
memcpy(&vect[-st],idata, (ENH_VECTL+st)*sizeof(float));
}
else {
en=st+ENH_VECTL;
if (en>idatal) {
memcpy(vect, &idata[st],
(ENH_VECTL-(en-idatal))*sizeof(float));
memset(&vect[ENH_VECTL-(en-idatal)], 0,
(en-idatal)*sizeof(float));
}
else {
memcpy(vect, &idata[st], ENH_VECTL*sizeof(float));
}
}
fraction=tloc2*ENH_UPS0-tloc;
/* compute the segment (this is actually a convolution) */
mycorr1(seg,vect,ENH_VECTL,polyphaserTbl+(2*ENH_FL0+1)*fraction,
2*ENH_FL0+1);
}
/*----------------------------------------------------------------*
* find the smoothed output data
*---------------------------------------------------------------*/
void smath(
float *odata, /* (o) smoothed output */
float *sseq,/* (i) said second sequence of waveforms */
int hl, /* (i) 2*hl+1 is sseq dimension */
float alpha0/* (i) max smoothing energy fraction */
){
int i,k;
float w00,w10,w11,A,B,C,*psseq,err,errs;
float surround[BLOCKL_MAX]; /* shape contributed by other than
current */
float wt[2*ENH_HL+1]; /* waveform weighting to get
surround shape */
float denom;
/* create shape of contribution from all waveforms except the
current one */
for (i=1; i<=2*hl+1; i++) {
wt[i-1] = (float)0.5*(1 - (float)cos(2*PI*i/(2*hl+2)));
}
wt[hl]=0.0; /* for clarity, not used */
for (i=0; i<ENH_BLOCKL; i++) {
surround[i]=sseq[i]*wt[0];
}
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for (k=1; k<hl; k++) {
psseq=sseq+k*ENH_BLOCKL;
for(i=0;i<ENH_BLOCKL; i++) {
surround[i]+=psseq[i]*wt[k];
}
}
for (k=hl+1; k<=2*hl; k++) {
psseq=sseq+k*ENH_BLOCKL;
for(i=0;i<ENH_BLOCKL; i++) {
surround[i]+=psseq[i]*wt[k];
}
}
/* compute some inner products */
w00 = w10 = w11 = 0.0;
psseq=sseq+hl*ENH_BLOCKL; /* current block */
for (i=0; i<ENH_BLOCKL;i++) {
w00+=psseq[i]*psseq[i];
w11+=surround[i]*surround[i];
w10+=surround[i]*psseq[i];
}
if (fabs(w11) < 1.0) {
w11=1.0;
}
C = (float)sqrt( w00/w11);
/* first try enhancement without power-constraint */
errs=0.0;
psseq=sseq+hl*ENH_BLOCKL;
for (i=0; i<ENH_BLOCKL; i++) {
odata[i]=C*surround[i];
err=psseq[i]-odata[i];
errs+=err*err;
}
/* if constraint violated by first try, add constraint */
if (errs > alpha0 * w00) {
if ( w00 < 1) {
w00=1;
}
denom = (w11*w00-w10*w10)/(w00*w00);
if (denom > 0.0001) { /* eliminates numerical problems
for if smooth */
A = (float)sqrt( (alpha0- alpha0*alpha0/4)/denom);
B = -alpha0/2 - A * w10/w00;
B = B+1;
}
else { /* essentially no difference between cycles;
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smoothing not needed */
A= 0.0;
B= 1.0;
}
/* create smoothed sequence */
psseq=sseq+hl*ENH_BLOCKL;
for (i=0; i<ENH_BLOCKL; i++) {
odata[i]=A*surround[i]+B*psseq[i];
}
}
}
/*----------------------------------------------------------------*
* get the pitch-synchronous sample sequence
*---------------------------------------------------------------*/
void getsseq(
float *sseq, /* (o) the pitch-synchronous sequence */
float *idata, /* (i) original data */
int idatal, /* (i) dimension of data */
int centerStartPos, /* (i) where current block starts */
float *period, /* (i) rough-pitch-period array */
float *plocs, /* (i) where periods of period array
are taken */
int periodl, /* (i) dimension period array */
int hl /* (i) 2*hl+1 is the number of sequences */
){
int i,centerEndPos,q;
float blockStartPos[2*ENH_HL+1];
int lagBlock[2*ENH_HL+1];
float plocs2[ENH_PLOCSL];
float *psseq;
centerEndPos=centerStartPos+ENH_BLOCKL-1;
/* present */
NearestNeighbor(lagBlock+hl,plocs,
(float)0.5*(centerStartPos+centerEndPos),periodl);
blockStartPos[hl]=(float)centerStartPos;
psseq=sseq+ENH_BLOCKL*hl;
memcpy(psseq, idata+centerStartPos, ENH_BLOCKL*sizeof(float));
/* past */
for (q=hl-1; q>=0; q--) {
blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
NearestNeighbor(lagBlock+q,plocs,
blockStartPos[q]+
ENH_BLOCKL_HALF-period[lagBlock[q+1]], periodl);
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if (blockStartPos[q]-ENH_OVERHANG>=0) {
refiner(sseq+q*ENH_BLOCKL, blockStartPos+q, idata,
idatal, centerStartPos, blockStartPos[q],
period[lagBlock[q+1]]);
} else {
psseq=sseq+q*ENH_BLOCKL;
memset(psseq, 0, ENH_BLOCKL*sizeof(float));
}
}
/* future */
for (i=0; i<periodl; i++) {
plocs2[i]=plocs[i]-period[i];
}
for (q=hl+1; q<=2*hl; q++) {
NearestNeighbor(lagBlock+q,plocs2,
blockStartPos[q-1]+ENH_BLOCKL_HALF,periodl);
blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
if (blockStartPos[q]+ENH_BLOCKL+ENH_OVERHANG<idatal) {
refiner(sseq+ENH_BLOCKL*q, blockStartPos+q, idata,
idatal, centerStartPos, blockStartPos[q],
period[lagBlock[q]]);
}
else {
psseq=sseq+q*ENH_BLOCKL;
memset(psseq, 0, ENH_BLOCKL*sizeof(float));
}
}
}
/*----------------------------------------------------------------*
* perform enhancement on idata+centerStartPos through
* idata+centerStartPos+ENH_BLOCKL-1
*---------------------------------------------------------------*/
void enhancer(
float *odata, /* (o) smoothed block, dimension blockl */
float *idata, /* (i) data buffer used for enhancing */
int idatal, /* (i) dimension idata */
int centerStartPos, /* (i) first sample current block
within idata */
float alpha0, /* (i) max correction-energy-fraction
(in [0,1]) */
float *period, /* (i) pitch period array */
float *plocs, /* (i) locations where period array
values valid */
int periodl /* (i) dimension of period and plocs */
){
float sseq[(2*ENH_HL+1)*ENH_BLOCKL];
/* get said second sequence of segments */
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getsseq(sseq,idata,idatal,centerStartPos,period,
plocs,periodl,ENH_HL);
/* compute the smoothed output from said second sequence */
smath(odata,sseq,ENH_HL,alpha0);
}
/*----------------------------------------------------------------*
* cross correlation
*---------------------------------------------------------------*/
float xCorrCoef(
float *target, /* (i) first array */
float *regressor, /* (i) second array */
int subl /* (i) dimension arrays */
){
int i;
float ftmp1, ftmp2;
ftmp1 = 0.0;
ftmp2 = 0.0;
for (i=0; i<subl; i++) {
ftmp1 += target[i]*regressor[i];
ftmp2 += regressor[i]*regressor[i];
}
if (ftmp1 > 0.0) {
return (float)(ftmp1*ftmp1/ftmp2);
}
else {
return (float)0.0;
}
}
/*----------------------------------------------------------------*
* interface for enhancer
*---------------------------------------------------------------*/
int enhancerInterface(
float *out, /* (o) enhanced signal */
float *in, /* (i) unenhanced signal */
iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
){
float *enh_buf, *enh_period;
int iblock, isample;
int lag=0, ilag, i, ioffset;
float cc, maxcc;
float ftmp1, ftmp2;
float *inPtr, *enh_bufPtr1, *enh_bufPtr2;
float plc_pred[ENH_BLOCKL];
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float lpState[6], downsampled[(ENH_NBLOCKS*ENH_BLOCKL+120)/2];
int inLen=ENH_NBLOCKS*ENH_BLOCKL+120;
int start, plc_blockl, inlag;
enh_buf=iLBCdec_inst->enh_buf;
enh_period=iLBCdec_inst->enh_period;
memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
(ENH_BUFL-iLBCdec_inst->blockl)*sizeof(float));
memcpy(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
iLBCdec_inst->blockl*sizeof(float));
if (iLBCdec_inst->mode==30)
plc_blockl=ENH_BLOCKL;
else
plc_blockl=40;
/* when 20 ms frame, move processing one block */
ioffset=0;
if (iLBCdec_inst->mode==20) ioffset=1;
i=3-ioffset;
memmove(enh_period, &enh_period[i],
(ENH_NBLOCKS_TOT-i)*sizeof(float));
/* Set state information to the 6 samples right before
the samples to be downsampled. */
memcpy(lpState,
enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-126,
6*sizeof(float));
/* Down sample a factor 2 to save computations */
DownSample(enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-120,
lpFilt_coefsTbl, inLen-ioffset*ENH_BLOCKL,
lpState, downsampled);
/* Estimate the pitch in the down sampled domain. */
for (iblock = 0; iblock<ENH_NBLOCKS-ioffset; iblock++) {
lag = 10;
maxcc = xCorrCoef(downsampled+60+iblock*
ENH_BLOCKL_HALF, downsampled+60+iblock*
ENH_BLOCKL_HALF-lag, ENH_BLOCKL_HALF);
for (ilag=11; ilag<60; ilag++) {
cc = xCorrCoef(downsampled+60+iblock*
ENH_BLOCKL_HALF, downsampled+60+iblock*
ENH_BLOCKL_HALF-ilag, ENH_BLOCKL_HALF);
if (cc > maxcc) {
maxcc = cc;
lag = ilag;
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}
}
/* Store the estimated lag in the non-downsampled domain */
enh_period[iblock+ENH_NBLOCKS_EXTRA+ioffset] = (float)lag*2;
}
/* PLC was performed on the previous packet */
if (iLBCdec_inst->prev_enh_pl==1) {
inlag=(int)enh_period[ENH_NBLOCKS_EXTRA+ioffset];
lag = inlag-1;
maxcc = xCorrCoef(in, in+lag, plc_blockl);
for (ilag=inlag; ilag<=inlag+1; ilag++) {
cc = xCorrCoef(in, in+ilag, plc_blockl);
if (cc > maxcc) {
maxcc = cc;
lag = ilag;
}
}
enh_period[ENH_NBLOCKS_EXTRA+ioffset-1]=(float)lag;
/* compute new concealed residual for the old lookahead,
mix the forward PLC with a backward PLC from
the new frame */
inPtr=&in[lag-1];
enh_bufPtr1=&plc_pred[plc_blockl-1];
if (lag>plc_blockl) {
start=plc_blockl;
} else {
start=lag;
}
for (isample = start; isample>0; isample--) {
*enh_bufPtr1-- = *inPtr--;
}
enh_bufPtr2=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
for (isample = (plc_blockl-1-lag); isample>=0; isample--) {
*enh_bufPtr1-- = *enh_bufPtr2--;
}
/* limit energy change */
ftmp2=0.0;
ftmp1=0.0;
for (i=0;i<plc_blockl;i++) {
ftmp2+=enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i]*
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enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i];
ftmp1+=plc_pred[i]*plc_pred[i];
}
ftmp1=(float)sqrt(ftmp1/(float)plc_blockl);
ftmp2=(float)sqrt(ftmp2/(float)plc_blockl);
if (ftmp1>(float)2.0*ftmp2 && ftmp1>0.0) {
for (i=0;i<plc_blockl-10;i++) {
plc_pred[i]*=(float)2.0*ftmp2/ftmp1;
}
for (i=plc_blockl-10;i<plc_blockl;i++) {
plc_pred[i]*=(float)(i-plc_blockl+10)*
((float)1.0-(float)2.0*ftmp2/ftmp1)/(float)(10)+
(float)2.0*ftmp2/ftmp1;
}
}
enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
for (i=0; i<plc_blockl; i++) {
ftmp1 = (float) (i+1) / (float) (plc_blockl+1);
*enh_bufPtr1 *= ftmp1;
*enh_bufPtr1 += ((float)1.0-ftmp1)*
plc_pred[plc_blockl-1-i];
enh_bufPtr1--;
}
}
if (iLBCdec_inst->mode==20) {
/* Enhancer with 40 samples delay */
for (iblock = 0; iblock<2; iblock++) {
enhancer(out+iblock*ENH_BLOCKL, enh_buf,
ENH_BUFL, (5+iblock)*ENH_BLOCKL+40,
ENH_ALPHA0, enh_period, enh_plocsTbl,
ENH_NBLOCKS_TOT);
}
} else if (iLBCdec_inst->mode==30) {
/* Enhancer with 80 samples delay */
for (iblock = 0; iblock<3; iblock++) {
enhancer(out+iblock*ENH_BLOCKL, enh_buf,
ENH_BUFL, (4+iblock)*ENH_BLOCKL,
ENH_ALPHA0, enh_period, enh_plocsTbl,
ENH_NBLOCKS_TOT);
}
}
return (lag*2);
}
A.17 filter.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
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filter.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_FILTER_H
#define __iLBC_FILTER_H
void AllPoleFilter(
float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
InOut[-1] contain the state of the
filter (delayed samples). InOut[0] to
InOut[lengthInOut-1] contain the filter
input, on en exit InOut[-orderCoef] to
InOut[-1] is unchanged and InOut[0] to
InOut[lengthInOut-1] contain filtered
samples */
float *Coef,/* (i) filter coefficients, Coef[0] is assumed
to be 1.0 */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef /* (i) number of filter coefficients */
);
void AllZeroFilter(
float *In, /* (i) In[0] to In[lengthInOut-1] contain
filter input samples */
float *Coef,/* (i) filter coefficients (Coef[0] is assumed
to be 1.0) */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef, /* (i) number of filter coefficients */
float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
contain the filter state, on exit Out[0]
to Out[lengthInOut-1] contain filtered
samples */
);
void ZeroPoleFilter(
float *In, /* (i) In[0] to In[lengthInOut-1] contain filter
input samples In[-orderCoef] to In[-1]
contain state of all-zero section */
float *ZeroCoef,/* (i) filter coefficients for all-zero
section (ZeroCoef[0] is assumed to
be 1.0) */
float *PoleCoef,/* (i) filter coefficients for all-pole section
(ZeroCoef[0] is assumed to be 1.0) */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef, /* (i) number of filter coefficients */
float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
contain state of all-pole section. On
exit Out[0] to Out[lengthInOut-1]
contain filtered samples */
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);
void DownSample (
float *In, /* (i) input samples */
float *Coef, /* (i) filter coefficients */
int lengthIn, /* (i) number of input samples */
float *state, /* (i) filter state */
float *Out /* (o) downsampled output */
);
#endif
A.18 filter.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
filter.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "iLBC_define.h"
/*----------------------------------------------------------------*
* all-pole filter
*---------------------------------------------------------------*/
void AllPoleFilter(
float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
InOut[-1] contain the state of the
filter (delayed samples). InOut[0] to
InOut[lengthInOut-1] contain the filter
input, on en exit InOut[-orderCoef] to
InOut[-1] is unchanged and InOut[0] to
InOut[lengthInOut-1] contain filtered
samples */
float *Coef,/* (i) filter coefficients, Coef[0] is assumed
to be 1.0 */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef /* (i) number of filter coefficients */
){
int n,k;
for(n=0;n<lengthInOut;n++){
for(k=1;k<=orderCoef;k++){
*InOut -= Coef[k]*InOut[-k];
}
InOut++;
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}
}
/*----------------------------------------------------------------*
* all-zero filter
*---------------------------------------------------------------*/
void AllZeroFilter(
float *In, /* (i) In[0] to In[lengthInOut-1] contain
filter input samples */
float *Coef,/* (i) filter coefficients (Coef[0] is assumed
to be 1.0) */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef, /* (i) number of filter coefficients */
float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
contain the filter state, on exit Out[0]
to Out[lengthInOut-1] contain filtered
samples */
){
int n,k;
for(n=0;n<lengthInOut;n++){
*Out = Coef[0]*In[0];
for(k=1;k<=orderCoef;k++){
*Out += Coef[k]*In[-k];
}
Out++;
In++;
}
}
/*----------------------------------------------------------------*
* pole-zero filter
*---------------------------------------------------------------*/
void ZeroPoleFilter(
float *In, /* (i) In[0] to In[lengthInOut-1] contain
filter input samples In[-orderCoef] to
In[-1] contain state of all-zero
section */
float *ZeroCoef,/* (i) filter coefficients for all-zero
section (ZeroCoef[0] is assumed to
be 1.0) */
float *PoleCoef,/* (i) filter coefficients for all-pole section
(ZeroCoef[0] is assumed to be 1.0) */
int lengthInOut,/* (i) number of input/output samples */
int orderCoef, /* (i) number of filter coefficients */
float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
contain state of all-pole section. On
exit Out[0] to Out[lengthInOut-1]
contain filtered samples */
){
AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
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AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
}
/*----------------------------------------------------------------*
* downsample (LP filter and decimation)
*---------------------------------------------------------------*/
void DownSample (
float *In, /* (i) input samples */
float *Coef, /* (i) filter coefficients */
int lengthIn, /* (i) number of input samples */
float *state, /* (i) filter state */
float *Out /* (o) downsampled output */
){
float o;
float *Out_ptr = Out;
float *Coef_ptr, *In_ptr;
float *state_ptr;
int i, j, stop;
/* LP filter and decimate at the same time */
for (i = DELAY_DS; i < lengthIn; i+=FACTOR_DS)
{
Coef_ptr = &Coef[0];
In_ptr = &In[i];
state_ptr = &state[FILTERORDER_DS-2];
o = (float)0.0;
stop = (i < FILTERORDER_DS) ? i + 1 : FILTERORDER_DS;
for (j = 0; j < stop; j++)
{
o += *Coef_ptr++ * (*In_ptr--);
}
for (j = i + 1; j < FILTERORDER_DS; j++)
{
o += *Coef_ptr++ * (*state_ptr--);
}
*Out_ptr++ = o;
}
/* Get the last part (use zeros as input for the future) */
for (i=(lengthIn+FACTOR_DS); i<(lengthIn+DELAY_DS);
i+=FACTOR_DS) {
o=(float)0.0;
if (i<lengthIn) {
Coef_ptr = &Coef[0];
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In_ptr = &In[i];
for (j=0; j<FILTERORDER_DS; j++) {
o += *Coef_ptr++ * (*Out_ptr--);
}
} else {
Coef_ptr = &Coef[i-lengthIn];
In_ptr = &In[lengthIn-1];
for (j=0; j<FILTERORDER_DS-(i-lengthIn); j++) {
o += *Coef_ptr++ * (*In_ptr--);
}
}
*Out_ptr++ = o;
}
}
A.19 FrameClassify.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
FrameClassify.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_FRAMECLASSIFY_H
#define __iLBC_FRAMECLASSIFY_H
int FrameClassify( /* index to the max-energy sub-frame */
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i/o) the encoder state structure */
float *residual /* (i) lpc residual signal */
);
#endif
A.20 FrameClassify.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
FrameClassify.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
Andersen et. al. Experimental - Expires April 25th, 2004 118
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******************************************************************/
#include "iLBC_define.h"
/*---------------------------------------------------------------*
* Classification of subframes to localize start state
*--------------------------------------------------------------*/
int FrameClassify( /* index to the max-energy sub-frame */
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i/o) the encoder state structure */
float *residual /* (i) lpc residual signal */
) {
float max_ssqEn, fssqEn[NSUB_MAX], bssqEn[NSUB_MAX], *pp;
int n, l, max_ssqEn_n;
const float ssqEn_win[NSUB_MAX-1]={(float)0.8,(float)0.9,
(float)1.0,(float)0.9,(float)0.8};
const float sampEn_win[5]={(float)1.0/(float)6.0,
(float)2.0/(float)6.0, (float)3.0/(float)6.0,
(float)4.0/(float)6.0, (float)5.0/(float)6.0};
/* init the front and back energies to zero */
memset(fssqEn, 0, NSUB_MAX*sizeof(float));
memset(bssqEn, 0, NSUB_MAX*sizeof(float));
/* Calculate front of first seqence */
n=0;
pp=residual;
for (l=0; l<5; l++) {
fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
pp++;
}
for (l=5; l<SUBL; l++) {
fssqEn[n] += (*pp) * (*pp);
pp++;
}
/* Calculate front and back of all middle sequences */
for (n=1; n<iLBCenc_inst->nsub-1; n++) {
pp=residual+n*SUBL;
for (l=0; l<5; l++) {
fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
bssqEn[n] += (*pp) * (*pp);
pp++;
}
for (l=5; l<SUBL-5; l++) {
fssqEn[n] += (*pp) * (*pp);
bssqEn[n] += (*pp) * (*pp);
pp++;
}
for (l=SUBL-5; l<SUBL; l++) {
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fssqEn[n] += (*pp) * (*pp);
bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
pp++;
}
}
/* Calculate back of last seqence */
n=iLBCenc_inst->nsub-1;
pp=residual+n*SUBL;
for (l=0; l<SUBL-5; l++) {
bssqEn[n] += (*pp) * (*pp);
pp++;
}
for (l=SUBL-5; l<SUBL; l++) {
bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
pp++;
}
/* find the index to the weighted 80 sample with
most energy */
if (iLBCenc_inst->mode==20) l=1;
else l=0;
max_ssqEn=(fssqEn[0]+bssqEn[1])*ssqEn_win[l];
max_ssqEn_n=1;
for (n=2; n<iLBCenc_inst->nsub; n++) {
l++;
if ((fssqEn[n-1]+bssqEn[n])*ssqEn_win[l] > max_ssqEn) {
max_ssqEn=(fssqEn[n-1]+bssqEn[n]) *
ssqEn_win[l];
max_ssqEn_n=n;
}
}
return max_ssqEn_n;
}
A.21 gainquant.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
gainquant.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
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#ifndef __iLBC_GAINQUANT_H
#define __iLBC_GAINQUANT_H
float gainquant(/* (o) quantized gain value */
float in, /* (i) gain value */
float maxIn,/* (i) maximum of gain value */
int cblen, /* (i) number of quantization indices */
int *index /* (o) quantization index */
);
float gaindequant( /* (o) quantized gain value */
int index, /* (i) quantization index */
float maxIn,/* (i) maximum of unquantized gain */
int cblen /* (i) number of quantization indices */
);
#endif
A.22 gainquant.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
gainquant.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <string.h>
#include <math.h>
#include "constants.h"
#include "filter.h"
/*----------------------------------------------------------------*
* quantizer for the gain in the gain-shape coding of residual
*---------------------------------------------------------------*/
float gainquant(/* (o) quantized gain value */
float in, /* (i) gain value */
float maxIn,/* (i) maximum of gain value */
int cblen, /* (i) number of quantization indices */
int *index /* (o) quantization index */
){
int i, tindex;
float minmeasure,measure, *cb, scale;
/* ensure a lower bound on the scaling factor */
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scale=maxIn;
if (scale<0.1) {
scale=(float)0.1;
}
/* select the quantization table */
if (cblen == 8) {
cb = gain_sq3Tbl;
} else if (cblen == 16) {
cb = gain_sq4Tbl;
} else {
cb = gain_sq5Tbl;
}
/* select the best index in the quantization table */
minmeasure=10000000.0;
tindex=0;
for (i=0; i<cblen; i++) {
measure=(in-scale*cb[i])*(in-scale*cb[i]);
if (measure<minmeasure) {
tindex=i;
minmeasure=measure;
}
}
*index=tindex;
/* return the quantized value */
return scale*cb[tindex];
}
/*----------------------------------------------------------------*
* decoder for quantized gains in the gain-shape coding of
* residual
*---------------------------------------------------------------*/
float gaindequant( /* (o) quantized gain value */
int index, /* (i) quantization index */
float maxIn,/* (i) maximum of unquantized gain */
int cblen /* (i) number of quantization indices */
){
float scale;
/* obtain correct scale factor */
scale=(float)fabs(maxIn);
if (scale<0.1) {
scale=(float)0.1;
}
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/* select the quantization table and return the decoded value */
if (cblen==8) {
return scale*gain_sq3Tbl[index];
} else if (cblen==16) {
return scale*gain_sq4Tbl[index];
}
else if (cblen==32) {
return scale*gain_sq5Tbl[index];
}
return 0.0;
}
A.23 getCBvec.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
getCBvec.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_GETCBVEC_H
#define __iLBC_GETCBVEC_H
void getCBvec(
float *cbvec, /* (o) Constructed codebook vector */
float *mem, /* (i) Codebook buffer */
int index, /* (i) Codebook index */
int lMem, /* (i) Length of codebook buffer */
int cbveclen/* (i) Codebook vector length */
);
#endif
A.24 getCBvec.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
getCBvec.c
Copyright (c) 2001-2003,
Global IP Sound AB.
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All rights reserved.
******************************************************************/
#include "iLBC_define.h"
#include "constants.h"
#include <string.h>
/*----------------------------------------------------------------*
* Construct codebook vector for given index.
*---------------------------------------------------------------*/
void getCBvec(
float *cbvec, /* (o) Constructed codebook vector */
float *mem, /* (i) Codebook buffer */
int index, /* (i) Codebook index */
int lMem, /* (i) Length of codebook buffer */
int cbveclen/* (i) Codebook vector length */
){
int j, k, n, memInd, sFilt;
float tmpbuf[CB_MEML];
int base_size;
int ilow, ihigh;
float alfa, alfa1;
/* Determine size of codebook sections */
base_size=lMem-cbveclen+1;
if (cbveclen==SUBL) {
base_size+=cbveclen/2;
}
/* No filter -> First codebook section */
if (index<lMem-cbveclen+1) {
/* first non-interpolated vectors */
k=index+cbveclen;
/* get vector */
memcpy(cbvec, mem+lMem-k, cbveclen*sizeof(float));
} else if (index < base_size) {
k=2*(index-(lMem-cbveclen+1))+cbveclen;
ihigh=k/2;
ilow=ihigh-5;
/* Copy first noninterpolated part */
memcpy(cbvec, mem+lMem-k/2, ilow*sizeof(float));
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/* interpolation */
alfa1=(float)0.2;
alfa=0.0;
for (j=ilow; j<ihigh; j++) {
cbvec[j]=((float)1.0-alfa)*mem[lMem-k/2+j]+
alfa*mem[lMem-k+j];
alfa+=alfa1;
}
/* Copy second noninterpolated part */
memcpy(cbvec+ihigh, mem+lMem-k+ihigh,
(cbveclen-ihigh)*sizeof(float));
}
/* Higher codebbok section based on filtering */
else {
/* first non-interpolated vectors */
if (index-base_size<lMem-cbveclen+1) {
float tempbuff2[CB_MEML+CB_FILTERLEN+1];
float *pos;
float *pp, *pp1;
memset(tempbuff2, 0,
CB_HALFFILTERLEN*sizeof(float));
memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
lMem*sizeof(float));
memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
(CB_HALFFILTERLEN+1)*sizeof(float));
k=index-base_size+cbveclen;
sFilt=lMem-k;
memInd=sFilt+1-CB_HALFFILTERLEN;
/* do filtering */
pos=cbvec;
memset(pos, 0, cbveclen*sizeof(float));
for (n=0; n<cbveclen; n++) {
pp=&tempbuff2[memInd+n+CB_HALFFILTERLEN];
pp1=&cbfiltersTbl[CB_FILTERLEN-1];
for (j=0; j<CB_FILTERLEN; j++) {
(*pos)+=(*pp++)*(*pp1--);
}
pos++;
}
}
/* interpolated vectors */
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else {
float tempbuff2[CB_MEML+CB_FILTERLEN+1];
float *pos;
float *pp, *pp1;
int i;
memset(tempbuff2, 0,
CB_HALFFILTERLEN*sizeof(float));
memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
lMem*sizeof(float));
memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
(CB_HALFFILTERLEN+1)*sizeof(float));
k=2*(index-base_size-
(lMem-cbveclen+1))+cbveclen;
sFilt=lMem-k;
memInd=sFilt+1-CB_HALFFILTERLEN;
/* do filtering */
pos=&tmpbuf[sFilt];
memset(pos, 0, k*sizeof(float));
for (i=0; i<k; i++) {
pp=&tempbuff2[memInd+i+CB_HALFFILTERLEN];
pp1=&cbfiltersTbl[CB_FILTERLEN-1];
for (j=0; j<CB_FILTERLEN; j++) {
(*pos)+=(*pp++)*(*pp1--);
}
pos++;
}
ihigh=k/2;
ilow=ihigh-5;
/* Copy first noninterpolated part */
memcpy(cbvec, tmpbuf+lMem-k/2,
ilow*sizeof(float));
/* interpolation */
alfa1=(float)0.2;
alfa=0.0;
for (j=ilow; j<ihigh; j++) {
cbvec[j]=((float)1.0-alfa)*
tmpbuf[lMem-k/2+j]+alfa*tmpbuf[lMem-k+j];
alfa+=alfa1;
}
/* Copy second noninterpolated part */
memcpy(cbvec+ihigh, tmpbuf+lMem-k+ihigh,
(cbveclen-ihigh)*sizeof(float));
}
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}
}
A.25 helpfun.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
helpfun.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_HELPFUN_H
#define __iLBC_HELPFUN_H
void autocorr(
float *r, /* (o) autocorrelation vector */
const float *x, /* (i) data vector */
int N, /* (i) length of data vector */
int order /* largest lag for calculated
autocorrelations */
);
void window(
float *z, /* (o) the windowed data */
const float *x, /* (i) the original data vector */
const float *y, /* (i) the window */
int N /* (i) length of all vectors */
);
void levdurb(
float *a, /* (o) lpc coefficient vector starting
with 1.0 */
float *k, /* (o) reflection coefficients */
float *r, /* (i) autocorrelation vector */
int order /* (i) order of lpc filter */
);
void interpolate(
float *out, /* (o) the interpolated vector */
float *in1, /* (i) the first vector for the
interpolation */
float *in2, /* (i) the second vector for the
interpolation */
float coef, /* (i) interpolation weights */
int length /* (i) length of all vectors */
);
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void bwexpand(
float *out, /* (o) the bandwidth expanded lpc
coefficients */
float *in, /* (i) the lpc coefficients before bandwidth
expansion */
float coef, /* (i) the bandwidth expansion factor */
int length /* (i) the length of lpc coefficient vectors */
);
void vq(
float *Xq, /* (o) the quantized vector */
int *index, /* (o) the quantization index */
const float *CB,/* (i) the vector quantization codebook */
float *X, /* (i) the vector to quantize */
int n_cb, /* (i) the number of vectors in the codebook */
int dim /* (i) the dimension of all vectors */
);
void SplitVQ(
float *qX, /* (o) the quantized vector */
int *index, /* (o) a vector of indexes for all vector
codebooks in the split */
float *X, /* (i) the vector to quantize */
const float *CB,/* (i) the quantizer codebook */
int nsplit, /* the number of vector splits */
const int *dim, /* the dimension of X and qX */
const int *cbsize /* the number of vectors in the codebook */
);
void sort_sq(
float *xq, /* (o) the quantized value */
int *index, /* (o) the quantization index */
float x, /* (i) the value to quantize */
const float *cb,/* (i) the quantization codebook */
int cb_size /* (i) the size of the quantization codebook */
);
int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
nonstable ones */
float *lsf, /* (i) a table of lsf vectors */
int dim, /* (i) the dimension of each lsf vector */
int NoAn /* (i) the number of lsf vectors in the
table */
);
#endif
A.26 helpfun.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
Andersen et. al. Experimental - Expires April 25th, 2004 128
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helpfun.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include "iLBC_define.h"
#include "constants.h"
/*----------------------------------------------------------------*
* calculation of auto correlation
*---------------------------------------------------------------*/
void autocorr(
float *r, /* (o) autocorrelation vector */
const float *x, /* (i) data vector */
int N, /* (i) length of data vector */
int order /* largest lag for calculated
autocorrelations */
){
int lag, n;
float sum;
for (lag = 0; lag <= order; lag++) {
sum = 0;
for (n = 0; n < N - lag; n++) {
sum += x[n] * x[n+lag];
}
r[lag] = sum;
}
}
/*----------------------------------------------------------------*
* window multiplication
*---------------------------------------------------------------*/
void window(
float *z, /* (o) the windowed data */
const float *x, /* (i) the original data vector */
const float *y, /* (i) the window */
int N /* (i) length of all vectors */
){
int i;
for (i = 0; i < N; i++) {
z[i] = x[i] * y[i];
}
}
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/*----------------------------------------------------------------*
* levinson-durbin solution for lpc coefficients
*---------------------------------------------------------------*/
void levdurb(
float *a, /* (o) lpc coefficient vector starting
with 1.0 */
float *k, /* (o) reflection coefficients */
float *r, /* (i) autocorrelation vector */
int order /* (i) order of lpc filter */
){
float sum, alpha;
int m, m_h, i;
a[0] = 1.0;
if (r[0] < EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
for (i = 0; i < order; i++) {
k[i] = 0;
a[i+1] = 0;
}
} else {
a[1] = k[0] = -r[1]/r[0];
alpha = r[0] + r[1] * k[0];
for (m = 1; m < order; m++){
sum = r[m + 1];
for (i = 0; i < m; i++){
sum += a[i+1] * r[m - i];
}
k[m] = -sum / alpha;
alpha += k[m] * sum;
m_h = (m + 1) >> 1;
for (i = 0; i < m_h; i++){
sum = a[i+1] + k[m] * a[m - i];
a[m - i] += k[m] * a[i+1];
a[i+1] = sum;
}
a[m+1] = k[m];
}
}
}
/*----------------------------------------------------------------*
* interpolation between vectors
*---------------------------------------------------------------*/
void interpolate(
float *out, /* (o) the interpolated vector */
float *in1, /* (i) the first vector for the
interpolation */
float *in2, /* (i) the second vector for the
interpolation */
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float coef, /* (i) interpolation weights */
int length /* (i) length of all vectors */
){
int i;
float invcoef;
invcoef = (float)1.0 - coef;
for (i = 0; i < length; i++) {
out[i] = coef * in1[i] + invcoef * in2[i];
}
}
/*----------------------------------------------------------------*
* lpc bandwidth expansion
*---------------------------------------------------------------*/
void bwexpand(
float *out, /* (o) the bandwidth expanded lpc
coefficients */
float *in, /* (i) the lpc coefficients before bandwidth
expansion */
float coef, /* (i) the bandwidth expansion factor */
int length /* (i) the length of lpc coefficient vectors */
){
int i;
float chirp;
chirp = coef;
out[0] = in[0];
for (i = 1; i < length; i++) {
out[i] = chirp * in[i];
chirp *= coef;
}
}
/*----------------------------------------------------------------*
* vector quantization
*---------------------------------------------------------------*/
void vq(
float *Xq, /* (o) the quantized vector */
int *index, /* (o) the quantization index */
const float *CB,/* (i) the vector quantization codebook */
float *X, /* (i) the vector to quantize */
int n_cb, /* (i) the number of vectors in the codebook */
int dim /* (i) the dimension of all vectors */
){
int i, j;
int pos, minindex;
float dist, tmp, mindist;
pos = 0;
mindist = FLOAT_MAX;
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minindex = 0;
for (j = 0; j < n_cb; j++) {
dist = X[0] - CB[pos];
dist *= dist;
for (i = 1; i < dim; i++) {
tmp = X[i] - CB[pos + i];
dist += tmp*tmp;
}
if (dist < mindist) {
mindist = dist;
minindex = j;
}
pos += dim;
}
for (i = 0; i < dim; i++) {
Xq[i] = CB[minindex*dim + i];
}
*index = minindex;
}
/*----------------------------------------------------------------*
* split vector quantization
*---------------------------------------------------------------*/
void SplitVQ(
float *qX, /* (o) the quantized vector */
int *index, /* (o) a vector of indexes for all vector
codebooks in the split */
float *X, /* (i) the vector to quantize */
const float *CB,/* (i) the quantizer codebook */
int nsplit, /* the number of vector splits */
const int *dim, /* the dimension of X and qX */
const int *cbsize /* the number of vectors in the codebook */
){
int cb_pos, X_pos, i;
cb_pos = 0;
X_pos= 0;
for (i = 0; i < nsplit; i++) {
vq(qX + X_pos, index + i, CB + cb_pos, X + X_pos,
cbsize[i], dim[i]);
X_pos += dim[i];
cb_pos += dim[i] * cbsize[i];
}
}
/*----------------------------------------------------------------*
* scalar quantization
*---------------------------------------------------------------*/
void sort_sq(
float *xq, /* (o) the quantized value */
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int *index, /* (o) the quantization index */
float x, /* (i) the value to quantize */
const float *cb,/* (i) the quantization codebook */
int cb_size /* (i) the size of the quantization codebook */
){
int i;
if (x <= cb[0]) {
*index = 0;
*xq = cb[0];
} else {
i = 0;
while ((x > cb[i]) && i < cb_size - 1) {
i++;
}
if (x > ((cb[i] + cb[i - 1])/2)) {
*index = i;
*xq = cb[i];
} else {
*index = i - 1;
*xq = cb[i - 1];
}
}
}
/*----------------------------------------------------------------*
* check for stability of lsf coefficients
*---------------------------------------------------------------*/
int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
nonstable ones */
float *lsf, /* (i) a table of lsf vectors */
int dim, /* (i) the dimension of each lsf vector */
int NoAn /* (i) the number of lsf vectors in the
table */
){
int k,n,m, Nit=2, change=0,pos;
float tmp;
static float eps=(float)0.039; /* 50 Hz */
static float eps2=(float)0.0195;
static float maxlsf=(float)3.14; /* 4000 Hz */
static float minlsf=(float)0.01; /* 0 Hz */
/* LSF separation check*/
for (n=0; n<Nit; n++) { /* Run through a couple of times */
for (m=0; m<NoAn; m++) { /* Number of analyses per frame */
for (k=0; k<(dim-1); k++) {
pos=m*dim+k;
if ((lsf[pos+1]-lsf[pos])<eps) {
if (lsf[pos+1]<lsf[pos]) {
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tmp=lsf[pos+1];
lsf[pos+1]= lsf[pos]+eps2;
lsf[pos]= lsf[pos+1]-eps2;
} else {
lsf[pos]-=eps2;
lsf[pos+1]+=eps2;
}
change=1;
}
if (lsf[pos]<minlsf) {
lsf[pos]=minlsf;
change=1;
}
if (lsf[pos]>maxlsf) {
lsf[pos]=maxlsf;
change=1;
}
}
}
}
return change;
}
A.27 hpInput.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
hpInput.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_HPINPUT_H
#define __iLBC_HPINPUT_H
void hpInput(
float *In, /* (i) vector to filter */
int len, /* (i) length of vector to filter */
float *Out, /* (o) the resulting filtered vector */
float *mem /* (i/o) the filter state */
);
#endif
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A.28 hpInput.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
hpInput.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "constants.h"
/*----------------------------------------------------------------*
* Input high-pass filter
*---------------------------------------------------------------*/
void hpInput(
float *In, /* (i) vector to filter */
int len, /* (i) length of vector to filter */
float *Out, /* (o) the resulting filtered vector */
float *mem /* (i/o) the filter state */
){
int i;
float *pi, *po;
/* all-zero section*/
pi = &In[0];
po = &Out[0];
for (i=0; i<len; i++) {
*po = hpi_zero_coefsTbl[0] * (*pi);
*po += hpi_zero_coefsTbl[1] * mem[0];
*po += hpi_zero_coefsTbl[2] * mem[1];
mem[1] = mem[0];
mem[0] = *pi;
po++;
pi++;
}
/* all-pole section*/
po = &Out[0];
for (i=0; i<len; i++) {
*po -= hpi_pole_coefsTbl[1] * mem[2];
*po -= hpi_pole_coefsTbl[2] * mem[3];
mem[3] = mem[2];
mem[2] = *po;
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po++;
}
}
A.29 hpOutput.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
hpOutput.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_HPOUTPUT_H
#define __iLBC_HPOUTPUT_H
void hpOutput(
float *In, /* (i) vector to filter */
int len,/* (i) length of vector to filter */
float *Out, /* (o) the resulting filtered vector */
float *mem /* (i/o) the filter state */
);
#endif
A.30 hpOutput.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
hpOutput.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "constants.h"
/*----------------------------------------------------------------*
* Output high-pass filter
*---------------------------------------------------------------*/
void hpOutput(
float *In, /* (i) vector to filter */
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int len,/* (i) length of vector to filter */
float *Out, /* (o) the resulting filtered vector */
float *mem /* (i/o) the filter state */
){
int i;
float *pi, *po;
/* all-zero section*/
pi = &In[0];
po = &Out[0];
for (i=0; i<len; i++) {
*po = hpo_zero_coefsTbl[0] * (*pi);
*po += hpo_zero_coefsTbl[1] * mem[0];
*po += hpo_zero_coefsTbl[2] * mem[1];
mem[1] = mem[0];
mem[0] = *pi;
po++;
pi++;
}
/* all-pole section*/
po = &Out[0];
for (i=0; i<len; i++) {
*po -= hpo_pole_coefsTbl[1] * mem[2];
*po -= hpo_pole_coefsTbl[2] * mem[3];
mem[3] = mem[2];
mem[2] = *po;
po++;
}
}
A.31 iCBConstruct.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iCBConstruct.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_ICBCONSTRUCT_H
#define __iLBC_ICBCONSTRUCT_H
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void index_conv_enc(
int *index /* (i/o) Codebook indexes */
);
void index_conv_dec(
int *index /* (i/o) Codebook indexes */
);
void iCBConstruct(
float *decvector, /* (o) Decoded vector */
int *index, /* (i) Codebook indices */
int *gain_index,/* (i) Gain quantization indices */
float *mem, /* (i) Buffer for codevector construction */
int lMem, /* (i) Length of buffer */
int veclen, /* (i) Length of vector */
int nStages /* (i) Number of codebook stages */
);
#endif
A.32 iCBConstruct.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iCBConstruct.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include "iLBC_define.h"
#include "gainquant.h"
#include "getCBvec.h"
/*----------------------------------------------------------------*
* Convert the codebook indexes to make the search easier
*---------------------------------------------------------------*/
void index_conv_enc(
int *index /* (i/o) Codebook indexes */
){
int k;
for (k=1; k<CB_NSTAGES; k++) {
if ((index[k]>=108)&&(index[k]<172)) {
index[k]-=64;
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} else if (index[k]>=236) {
index[k]-=128;
} else {
/* ERROR */
}
}
}
void index_conv_dec(
int *index /* (i/o) Codebook indexes */
){
int k;
for (k=1; k<CB_NSTAGES; k++) {
if ((index[k]>=44)&&(index[k]<108)) {
index[k]+=64;
} else if ((index[k]>=108)&&(index[k]<128)) {
index[k]+=128;
} else {
/* ERROR */
}
}
}
/*----------------------------------------------------------------*
* Construct decoded vector from codebook and gains.
*---------------------------------------------------------------*/
void iCBConstruct(
float *decvector, /* (o) Decoded vector */
int *index, /* (i) Codebook indices */
int *gain_index,/* (i) Gain quantization indices */
float *mem, /* (i) Buffer for codevector construction */
int lMem, /* (i) Length of buffer */
int veclen, /* (i) Length of vector */
int nStages /* (i) Number of codebook stages */
){
int j,k;
float gain[CB_NSTAGES];
float cbvec[SUBL];
/* gain de-quantization */
gain[0] = gaindequant(gain_index[0], 1.0, 32);
if (nStages > 1) {
gain[1] = gaindequant(gain_index[1],
(float)fabs(gain[0]), 16);
}
if (nStages > 2) {
gain[2] = gaindequant(gain_index[2],
(float)fabs(gain[1]), 8);
}
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/* codebook vector construction and construction of
total vector */
getCBvec(cbvec, mem, index[0], lMem, veclen);
for (j=0;j<veclen;j++){
decvector[j] = gain[0]*cbvec[j];
}
if (nStages > 1) {
for (k=1; k<nStages; k++) {
getCBvec(cbvec, mem, index[k], lMem, veclen);
for (j=0;j<veclen;j++) {
decvector[j] += gain[k]*cbvec[j];
}
}
}
}
A.33 iCBSearch.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iCBSearch.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_ICBSEARCH_H
#define __iLBC_ICBSEARCH_H
void iCBSearch(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) the encoder state structure */
int *index, /* (o) Codebook indices */
int *gain_index,/* (o) Gain quantization indices */
float *intarget,/* (i) Target vector for encoding */
float *mem, /* (i) Buffer for codebook construction */
int lMem, /* (i) Length of buffer */
int lTarget, /* (i) Length of vector */
int nStages, /* (i) Number of codebook stages */
float *weightDenum, /* (i) weighting filter coefficients */
float *weightState, /* (i) weighting filter state */
int block /* (i) the sub-block number */
);
#endif
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A.34 iCBSearch.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
iCBSearch.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <string.h>
#include "iLBC_define.h"
#include "gainquant.h"
#include "createCB.h"
#include "filter.h"
#include "constants.h"
/*----------------------------------------------------------------*
* Search routine for codebook encoding and gain quantization.
*---------------------------------------------------------------*/
void iCBSearch(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) the encoder state structure */
int *index, /* (o) Codebook indices */
int *gain_index,/* (o) Gain quantization indices */
float *intarget,/* (i) Target vector for encoding */
float *mem, /* (i) Buffer for codebook construction */
int lMem, /* (i) Length of buffer */
int lTarget, /* (i) Length of vector */
int nStages, /* (i) Number of codebook stages */
float *weightDenum, /* (i) weighting filter coefficients */
float *weightState, /* (i) weighting filter state */
int block /* (i) the sub-block number */
){
int i, j, icount, stage, best_index, range, counter;
float max_measure, gain, measure, crossDot, ftmp;
float gains[CB_NSTAGES];
float target[SUBL];
int base_index, sInd, eInd, base_size;
int sIndAug=0, eIndAug=0;
float buf[CB_MEML+SUBL+2*LPC_FILTERORDER];
float invenergy[CB_EXPAND*128], energy[CB_EXPAND*128];
float *pp, *ppi=0, *ppo=0, *ppe=0;
float cbvectors[CB_MEML];
float tene, cene, cvec[SUBL];
float aug_vec[SUBL];
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memset(cvec,0,SUBL*sizeof(float));
/* Determine size of codebook sections */
base_size=lMem-lTarget+1;
if (lTarget==SUBL) {
base_size=lMem-lTarget+1+lTarget/2;
}
/* setup buffer for weighting */
memcpy(buf,weightState,sizeof(float)*LPC_FILTERORDER);
memcpy(buf+LPC_FILTERORDER,mem,lMem*sizeof(float));
memcpy(buf+LPC_FILTERORDER+lMem,intarget,lTarget*sizeof(float));
/* weighting */
AllPoleFilter(buf+LPC_FILTERORDER, weightDenum,
lMem+lTarget, LPC_FILTERORDER);
/* Construct the codebook and target needed */
memcpy(target, buf+LPC_FILTERORDER+lMem, lTarget*sizeof(float));
tene=0.0;
for (i=0; i<lTarget; i++) {
tene+=target[i]*target[i];
}
/* Prepare search over one more codebook section. This section
is created by filtering the original buffer with a filter. */
filteredCBvecs(cbvectors, buf+LPC_FILTERORDER, lMem);
/* The Main Loop over stages */
for (stage=0; stage<nStages; stage++) {
range = search_rangeTbl[block][stage];
/* initialize search measure */
max_measure = (float)-10000000.0;
gain = (float)0.0;
best_index = 0;
/* Compute cross dot product between the target
and the CB memory */
crossDot=0.0;
pp=buf+LPC_FILTERORDER+lMem-lTarget;
for (j=0; j<lTarget; j++) {
crossDot += target[j]*(*pp++);
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}
if (stage==0) {
/* Calculate energy in the first block of
'lTarget' sampels. */
ppe = energy;
ppi = buf+LPC_FILTERORDER+lMem-lTarget-1;
ppo = buf+LPC_FILTERORDER+lMem-1;
*ppe=0.0;
pp=buf+LPC_FILTERORDER+lMem-lTarget;
for (j=0; j<lTarget; j++) {
*ppe+=(*pp)*(*pp++);
}
if (*ppe>0.0) {
invenergy[0] = (float) 1.0 / (*ppe + EPS);
} else {
invenergy[0] = (float) 0.0;
}
ppe++;
measure=(float)-10000000.0;
if (crossDot > 0.0) {
measure = crossDot*crossDot*invenergy[0];
}
}
else {
measure = crossDot*crossDot*invenergy[0];
}
/* check if measure is better */
ftmp = crossDot*invenergy[0];
if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
best_index = 0;
max_measure = measure;
gain = ftmp;
}
/* loop over the main first codebook section,
full search */
for (icount=1; icount<range; icount++) {
/* calculate measure */
crossDot=0.0;
pp = buf+LPC_FILTERORDER+lMem-lTarget-icount;
for (j=0; j<lTarget; j++) {
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crossDot += target[j]*(*pp++);
}
if (stage==0) {
*ppe++ = energy[icount-1] + (*ppi)*(*ppi) -
(*ppo)*(*ppo);
ppo--;
ppi--;
if (energy[icount]>0.0) {
invenergy[icount] =
(float)1.0/(energy[icount]+EPS);
} else {
invenergy[icount] = (float) 0.0;
}
measure=(float)-10000000.0;
if (crossDot > 0.0) {
measure = crossDot*crossDot*invenergy[icount];
}
}
else {
measure = crossDot*crossDot*invenergy[icount];
}
/* check if measure is better */
ftmp = crossDot*invenergy[icount];
if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
best_index = icount;
max_measure = measure;
gain = ftmp;
}
}
/* Loop over augmented part in the first codebook
* section, full search.
* The vectors are interpolated.
*/
if (lTarget==SUBL) {
/* Search for best possible cb vector and
compute the CB-vectors' energy. */
searchAugmentedCB(20, 39, stage, base_size-lTarget/2,
target, buf+LPC_FILTERORDER+lMem,
&max_measure, &best_index, &gain, energy,
invenergy);
}
/* set search range for following codebook sections */
base_index=best_index;
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/* unrestricted search */
if (CB_RESRANGE == -1) {
sInd=0;
eInd=range-1;
sIndAug=20;
eIndAug=39;
}
/* restriced search around best index from first
codebook section */
else {
/* Initialize search indices */
sIndAug=0;
eIndAug=0;
sInd=base_index-CB_RESRANGE/2;
eInd=sInd+CB_RESRANGE;
if (lTarget==SUBL) {
if (sInd<0) {
sIndAug = 40 + sInd;
eIndAug = 39;
sInd=0;
} else if ( base_index < (base_size-20) ) {
if (eInd > range) {
sInd -= (eInd-range);
eInd = range;
}
} else { /* base_index >= (base_size-20) */
if (sInd < (base_size-20)) {
sIndAug = 20;
sInd = 0;
eInd = 0;
eIndAug = 19 + CB_RESRANGE;
if(eIndAug > 39) {
eInd = eIndAug-39;
eIndAug = 39;
}
} else {
sIndAug = 20 + sInd - (base_size-20);
eIndAug = 39;
sInd = 0;
eInd = CB_RESRANGE - (eIndAug-sIndAug+1);
}
}
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} else { /* lTarget = 22 or 23 */
if (sInd < 0) {
eInd -= sInd;
sInd = 0;
}
if(eInd > range) {
sInd -= (eInd - range);
eInd = range;
}
}
}
/* search of higher codebook section */
/* index search range */
counter = sInd;
sInd += base_size;
eInd += base_size;
if (stage==0) {
ppe = energy+base_size;
*ppe=0.0;
pp=cbvectors+lMem-lTarget;
for (j=0; j<lTarget; j++) {
*ppe+=(*pp)*(*pp++);
}
ppi = cbvectors + lMem - 1 - lTarget;
ppo = cbvectors + lMem - 1;
for (j=0; j<(range-1); j++) {
*(ppe+1) = *ppe + (*ppi)*(*ppi) - (*ppo)*(*ppo);
ppo--;
ppi--;
ppe++;
}
}
/* loop over search range */
for (icount=sInd; icount<eInd; icount++) {
/* calculate measure */
crossDot=0.0;
pp=cbvectors + lMem - (counter++) - lTarget;
for (j=0;j<lTarget;j++) {
crossDot += target[j]*(*pp++);
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}
if (energy[icount]>0.0) {
invenergy[icount] =(float)1.0/(energy[icount]+EPS);
} else {
invenergy[icount] =(float)0.0;
}
if (stage==0) {
measure=(float)-10000000.0;
if (crossDot > 0.0) {
measure = crossDot*crossDot*
invenergy[icount];
}
}
else {
measure = crossDot*crossDot*invenergy[icount];
}
/* check if measure is better */
ftmp = crossDot*invenergy[icount];
if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
best_index = icount;
max_measure = measure;
gain = ftmp;
}
}
/* Search the augmented CB inside the limited range. */
if ((lTarget==SUBL)&&(sIndAug!=0)) {
searchAugmentedCB(sIndAug, eIndAug, stage,
2*base_size-20, target, cbvectors+lMem,
&max_measure, &best_index, &gain, energy,
invenergy);
}
/* record best index */
index[stage] = best_index;
/* gain quantization */
if (stage==0){
if (gain<0.0){
gain = 0.0;
}
if (gain>CB_MAXGAIN) {
gain = (float)CB_MAXGAIN;
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}
gain = gainquant(gain, 1.0, 32, &gain_index[stage]);
}
else {
if (stage==1) {
gain = gainquant(gain, (float)fabs(gains[stage-1]),
16, &gain_index[stage]);
} else {
gain = gainquant(gain, (float)fabs(gains[stage-1]),
8, &gain_index[stage]);
}
}
/* Extract the best (according to measure)
codebook vector */
if (lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
if (index[stage]<base_size) {
pp=buf+LPC_FILTERORDER+lMem-lTarget-index[stage];
} else {
pp=cbvectors+lMem-lTarget-
index[stage]+base_size;
}
} else {
if (index[stage]<base_size) {
if (index[stage]<(base_size-20)) {
pp=buf+LPC_FILTERORDER+lMem-
lTarget-index[stage];
} else {
createAugmentedVec(index[stage]-base_size+40,
buf+LPC_FILTERORDER+lMem,aug_vec);
pp=aug_vec;
}
} else {
int filterno, position;
filterno=index[stage]/base_size;
position=index[stage]-filterno*base_size;
if (position<(base_size-20)) {
pp=cbvectors+filterno*lMem-lTarget-
index[stage]+filterno*base_size;
} else {
createAugmentedVec(
index[stage]-(filterno+1)*base_size+40,
cbvectors+filterno*lMem,aug_vec);
pp=aug_vec;
}
}
}
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/* Subtract the best codebook vector, according
to measure, from the target vector */
for (j=0;j<lTarget;j++) {
cvec[j] += gain*(*pp);
target[j] -= gain*(*pp++);
}
/* record quantized gain */
gains[stage]=gain;
}/* end of Main Loop. for (stage=0;... */
/* Gain adjustment for energy matching */
cene=0.0;
for (i=0; i<lTarget; i++) {
cene+=cvec[i]*cvec[i];
}
j=gain_index[0];
for (i=gain_index[0]; i<32; i++) {
ftmp=cene*gain_sq5Tbl[i]*gain_sq5Tbl[i];
if ((ftmp<(tene*gains[0]*gains[0])) &&
(gain_sq5Tbl[j]<(2.0*gains[0]))) {
j=i;
}
}
gain_index[0]=j;
}
A.35 LPCdecode.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
LPC_decode.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_LPC_DECODE_H
#define __iLBC_LPC_DECODE_H
void LSFinterpolate2a_dec(
float *a, /* (o) lpc coefficients for a sub-frame */
float *lsf1, /* (i) first lsf coefficient vector */
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float *lsf2, /* (i) second lsf coefficient vector */
float coef, /* (i) interpolation weight */
int length /* (i) length of lsf vectors */
);
void SimplelsfDEQ(
float *lsfdeq, /* (o) dequantized lsf coefficients */
int *index, /* (i) quantization index */
int lpc_n /* (i) number of LPCs */
);
void DecoderInterpolateLSF(
float *syntdenum, /* (o) synthesis filter coefficients */
float *weightdenum, /* (o) weighting denumerator
coefficients */
float *lsfdeq, /* (i) dequantized lsf coefficients */
int length, /* (i) length of lsf coefficient vector */
iLBC_Dec_Inst_t *iLBCdec_inst
/* (i) the decoder state structure */
);
#endif
A.36 LPCdecode.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
LPC_decode.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <string.h>
#include "helpfun.h"
#include "lsf.h"
#include "iLBC_define.h"
#include "constants.h"
/*---------------------------------------------------------------*
* interpolation of lsf coefficients for the decoder
*--------------------------------------------------------------*/
void LSFinterpolate2a_dec(
float *a, /* (o) lpc coefficients for a sub-frame */
float *lsf1, /* (i) first lsf coefficient vector */
float *lsf2, /* (i) second lsf coefficient vector */
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float coef, /* (i) interpolation weight */
int length /* (i) length of lsf vectors */
){
float lsftmp[LPC_FILTERORDER];
interpolate(lsftmp, lsf1, lsf2, coef, length);
lsf2a(a, lsftmp);
}
/*---------------------------------------------------------------*
* obtain dequantized lsf coefficients from quantization index
*--------------------------------------------------------------*/
void SimplelsfDEQ(
float *lsfdeq, /* (o) dequantized lsf coefficients */
int *index, /* (i) quantization index */
int lpc_n /* (i) number of LPCs */
){
int i, j, pos, cb_pos;
/* decode first LSF */
pos = 0;
cb_pos = 0;
for (i = 0; i < LSF_NSPLIT; i++) {
for (j = 0; j < dim_lsfCbTbl[i]; j++) {
lsfdeq[pos + j] = lsfCbTbl[cb_pos +
(long)(index[i])*dim_lsfCbTbl[i] + j];
}
pos += dim_lsfCbTbl[i];
cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
}
if (lpc_n>1) {
/* decode last LSF */
pos = 0;
cb_pos = 0;
for (i = 0; i < LSF_NSPLIT; i++) {
for (j = 0; j < dim_lsfCbTbl[i]; j++) {
lsfdeq[LPC_FILTERORDER + pos + j] =
lsfCbTbl[cb_pos +
(long)(index[LSF_NSPLIT + i])*
dim_lsfCbTbl[i] + j];
}
pos += dim_lsfCbTbl[i];
cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
}
}
}
/*----------------------------------------------------------------*
* obtain synthesis and weighting filters form lsf coefficients
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*---------------------------------------------------------------*/
void DecoderInterpolateLSF(
float *syntdenum, /* (o) synthesis filter coefficients */
float *weightdenum, /* (o) weighting denumerator
coefficients */
float *lsfdeq, /* (i) dequantized lsf coefficients */
int length, /* (i) length of lsf coefficient vector */
iLBC_Dec_Inst_t *iLBCdec_inst
/* (i) the decoder state structure */
){
int i, pos, lp_length;
float lp[LPC_FILTERORDER + 1], *lsfdeq2;
lsfdeq2 = lsfdeq + length;
lp_length = length + 1;
if (iLBCdec_inst->mode==30) {
/* sub-frame 1: Interpolation between old and first */
LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
lsf_weightTbl_30ms[0], length);
memcpy(syntdenum,lp,lp_length*sizeof(float));
bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM,
lp_length);
/* sub-frames 2 to 6: interpolation between first
and last LSF */
pos = lp_length;
for (i = 1; i < 6; i++) {
LSFinterpolate2a_dec(lp, lsfdeq, lsfdeq2,
lsf_weightTbl_30ms[i], length);
memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
bwexpand(weightdenum + pos, lp,
LPC_CHIRP_WEIGHTDENUM, lp_length);
pos += lp_length;
}
}
else {
pos = 0;
for (i = 0; i < iLBCdec_inst->nsub; i++) {
LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold,
lsfdeq, lsf_weightTbl_20ms[i], length);
memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
bwexpand(weightdenum+pos, lp, LPC_CHIRP_WEIGHTDENUM,
lp_length);
pos += lp_length;
}
}
/* update memory */
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if (iLBCdec_inst->mode==30)
memcpy(iLBCdec_inst->lsfdeqold, lsfdeq2,
length*sizeof(float));
else
memcpy(iLBCdec_inst->lsfdeqold, lsfdeq,
length*sizeof(float));
}
A.37 LPCencode.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
LPCencode.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_LPCENCOD_H
#define __iLBC_LPCENCOD_H
void LPCencode(
float *syntdenum, /* (i/o) synthesis filter coefficients
before/after encoding */
float *weightdenum, /* (i/o) weighting denumerator coefficients
before/after encoding */
int *lsf_index, /* (o) lsf quantization index */
float *data, /* (i) lsf coefficients to quantize */
iLBC_Enc_Inst_t *iLBCenc_inst
/* (i/o) the encoder state structure */
);
#endif
A.38 LPCencode.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
LPCencode.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
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#include <string.h>
#include "iLBC_define.h"
#include "helpfun.h"
#include "lsf.h"
#include "constants.h"
/*----------------------------------------------------------------*
* lpc analysis (subrutine to LPCencode)
*---------------------------------------------------------------*/
void SimpleAnalysis(
float *lsf, /* (o) lsf coefficients */
float *data, /* (i) new data vector */
iLBC_Enc_Inst_t *iLBCenc_inst
/* (i/o) the encoder state structure */
){
int k, is;
float temp[BLOCKL_MAX], lp[LPC_FILTERORDER + 1];
float lp2[LPC_FILTERORDER + 1];
float r[LPC_FILTERORDER + 1];
is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
memcpy(iLBCenc_inst->lpc_buffer+is,data,
iLBCenc_inst->blockl*sizeof(float));
/* No lookahead, last window is asymmetric */
for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
is = LPC_LOOKBACK;
if (k < (iLBCenc_inst->lpc_n - 1)) {
window(temp, lpc_winTbl,
iLBCenc_inst->lpc_buffer, BLOCKL_MAX);
} else {
window(temp, lpc_asymwinTbl,
iLBCenc_inst->lpc_buffer + is, BLOCKL_MAX);
}
autocorr(r, temp, BLOCKL_MAX, LPC_FILTERORDER);
window(r, r, lpc_lagwinTbl, LPC_FILTERORDER + 1);
levdurb(lp, temp, r, LPC_FILTERORDER);
bwexpand(lp2, lp, LPC_CHIRP_SYNTDENUM, LPC_FILTERORDER+1);
a2lsf(lsf + k*LPC_FILTERORDER, lp2);
}
is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
memmove(iLBCenc_inst->lpc_buffer,
iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is,
is*sizeof(float));
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}
/*----------------------------------------------------------------*
* lsf interpolator and conversion from lsf to a coefficients
* (subrutine to SimpleInterpolateLSF)
*---------------------------------------------------------------*/
void LSFinterpolate2a_enc(
float *a, /* (o) lpc coefficients */
float *lsf1,/* (i) first set of lsf coefficients */
float *lsf2,/* (i) second set of lsf coefficients */
float coef, /* (i) weighting coefficient to use between
lsf1 and lsf2 */
long length /* (i) length of coefficient vectors */
){
float lsftmp[LPC_FILTERORDER];
interpolate(lsftmp, lsf1, lsf2, coef, length);
lsf2a(a, lsftmp);
}
/*----------------------------------------------------------------*
* lsf interpolator (subrutine to LPCencode)
*---------------------------------------------------------------*/
void SimpleInterpolateLSF(
float *syntdenum, /* (o) the synthesis filter denominator
resulting from the quantized
interpolated lsf */
float *weightdenum, /* (o) the weighting filter denominator
resulting from the unquantized
interpolated lsf */
float *lsf, /* (i) the unquantized lsf coefficients */
float *lsfdeq, /* (i) the dequantized lsf coefficients */
float *lsfold, /* (i) the unquantized lsf coefficients of
the previous signal frame */
float *lsfdeqold, /* (i) the dequantized lsf coefficients of
the previous signal frame */
int length, /* (i) should equate LPC_FILTERORDER */
iLBC_Enc_Inst_t *iLBCenc_inst
/* (i/o) the encoder state structure */
){
int i, pos, lp_length;
float lp[LPC_FILTERORDER + 1], *lsf2, *lsfdeq2;
lsf2 = lsf + length;
lsfdeq2 = lsfdeq + length;
lp_length = length + 1;
if (iLBCenc_inst->mode==30) {
/* sub-frame 1: Interpolation between old and first
set of lsf coefficients */
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LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
lsf_weightTbl_30ms[0], length);
memcpy(syntdenum,lp,lp_length*sizeof(float));
LSFinterpolate2a_enc(lp, lsfold, lsf,
lsf_weightTbl_30ms[0], length);
bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM, lp_length);
/* sub-frame 2 to 6: Interpolation between first
and second set of lsf coefficients */
pos = lp_length;
for (i = 1; i < iLBCenc_inst->nsub; i++) {
LSFinterpolate2a_enc(lp, lsfdeq, lsfdeq2,
lsf_weightTbl_30ms[i], length);
memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
LSFinterpolate2a_enc(lp, lsf, lsf2,
lsf_weightTbl_30ms[i], length);
bwexpand(weightdenum + pos, lp,
LPC_CHIRP_WEIGHTDENUM, lp_length);
pos += lp_length;
}
}
else {
pos = 0;
for (i = 0; i < iLBCenc_inst->nsub; i++) {
LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
lsf_weightTbl_20ms[i], length);
memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
LSFinterpolate2a_enc(lp, lsfold, lsf,
lsf_weightTbl_20ms[i], length);
bwexpand(weightdenum+pos, lp,
LPC_CHIRP_WEIGHTDENUM, lp_length);
pos += lp_length;
}
}
/* update memory */
if (iLBCenc_inst->mode==30) {
memcpy(lsfold, lsf2, length*sizeof(float));
memcpy(lsfdeqold, lsfdeq2, length*sizeof(float));
}
else {
memcpy(lsfold, lsf, length*sizeof(float));
memcpy(lsfdeqold, lsfdeq, length*sizeof(float));
}
}
/*----------------------------------------------------------------*
* lsf quantizer (subrutine to LPCencode)
*---------------------------------------------------------------*/
void SimplelsfQ(
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float *lsfdeq, /* (o) dequantized lsf coefficients
(dimension FILTERORDER) */
int *index, /* (o) quantization index */
float *lsf, /* (i) the lsf coefficient vector to be
quantized (dimension FILTERORDER ) */
int lpc_n /* (i) number of lsf sets to quantize */
){
/* Quantize first LSF with memoryless split VQ */
SplitVQ(lsfdeq, index, lsf, lsfCbTbl, LSF_NSPLIT,
dim_lsfCbTbl, size_lsfCbTbl);
if (lpc_n==2) {
/* Quantize second LSF with memoryless split VQ */
SplitVQ(lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
lsf + LPC_FILTERORDER, lsfCbTbl, LSF_NSPLIT,
dim_lsfCbTbl, size_lsfCbTbl);
}
}
/*----------------------------------------------------------------*
* lpc encoder
*---------------------------------------------------------------*/
void LPCencode(
float *syntdenum, /* (i/o) synthesis filter coefficients
before/after encoding */
float *weightdenum, /* (i/o) weighting denumerator
coefficients before/after
encoding */
int *lsf_index, /* (o) lsf quantization index */
float *data, /* (i) lsf coefficients to quantize */
iLBC_Enc_Inst_t *iLBCenc_inst
/* (i/o) the encoder state structure */
){
float lsf[LPC_FILTERORDER * LPC_N_MAX];
float lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
int change=0;
SimpleAnalysis(lsf, data, iLBCenc_inst);
SimplelsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
change=LSF_check(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
SimpleInterpolateLSF(syntdenum, weightdenum,
lsf, lsfdeq, iLBCenc_inst->lsfold,
iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
}
A.39 lsf.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
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lsf.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_LSF_H
#define __iLBC_LSF_H
void a2lsf(
float *freq,/* (o) lsf coefficients */
float *a /* (i) lpc coefficients */
);
void lsf2a(
float *a_coef, /* (o) lpc coefficients */
float *freq /* (i) lsf coefficients */
);
#endif
A.40 lsf.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
lsf.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <string.h>
#include <math.h>
#include "iLBC_define.h"
/*----------------------------------------------------------------*
* conversion from lpc coefficients to lsf coefficients
*---------------------------------------------------------------*/
void a2lsf(
float *freq,/* (o) lsf coefficients */
float *a /* (i) lpc coefficients */
){
float steps[LSF_NUMBER_OF_STEPS] =
{(float)0.00635, (float)0.003175, (float)0.0015875,
(float)0.00079375};
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float step;
int step_idx;
int lsp_index;
float p[LPC_HALFORDER];
float q[LPC_HALFORDER];
float p_pre[LPC_HALFORDER];
float q_pre[LPC_HALFORDER];
float old_p, old_q, *old;
float *pq_coef;
float omega, old_omega;
int i;
float hlp, hlp1, hlp2, hlp3, hlp4, hlp5;
for (i=0; i<LPC_HALFORDER; i++) {
p[i] = (float)-1.0 * (a[i + 1] + a[LPC_FILTERORDER - i]);
q[i] = a[LPC_FILTERORDER - i] - a[i + 1];
}
p_pre[0] = (float)-1.0 - p[0];
p_pre[1] = - p_pre[0] - p[1];
p_pre[2] = - p_pre[1] - p[2];
p_pre[3] = - p_pre[2] - p[3];
p_pre[4] = - p_pre[3] - p[4];
p_pre[4] = p_pre[4] / 2;
q_pre[0] = (float)1.0 - q[0];
q_pre[1] = q_pre[0] - q[1];
q_pre[2] = q_pre[1] - q[2];
q_pre[3] = q_pre[2] - q[3];
q_pre[4] = q_pre[3] - q[4];
q_pre[4] = q_pre[4] / 2;
omega = 0.0;
old_omega = 0.0;
old_p = FLOAT_MAX;
old_q = FLOAT_MAX;
/* Here we loop through lsp_index to find all the
LPC_FILTERORDER roots for omega. */
for (lsp_index = 0; lsp_index<LPC_FILTERORDER; lsp_index++) {
/* Depending on lsp_index being even or odd, we
alternatively solve the roots for the two LSP equations. */
if ((lsp_index & 0x1) == 0) {
pq_coef = p_pre;
old = &old_p;
} else {
pq_coef = q_pre;
old = &old_q;
}
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/* Start with low resolution grid */
for (step_idx = 0, step = steps[step_idx];
step_idx < LSF_NUMBER_OF_STEPS;){
/* cos(10piw) + pq(0)cos(8piw) + pq(1)cos(6piw) +
pq(2)cos(4piw) + pq(3)cod(2piw) + pq(4) */
hlp = (float)cos(omega * TWO_PI);
hlp1 = (float)2.0 * hlp + pq_coef[0];
hlp2 = (float)2.0 * hlp * hlp1 - (float)1.0 +
pq_coef[1];
hlp3 = (float)2.0 * hlp * hlp2 - hlp1 + pq_coef[2];
hlp4 = (float)2.0 * hlp * hlp3 - hlp2 + pq_coef[3];
hlp5 = hlp * hlp4 - hlp3 + pq_coef[4];
if (((hlp5 * (*old)) <= 0.0) || (omega >= 0.5)){
if (step_idx == (LSF_NUMBER_OF_STEPS - 1)){
if (fabs(hlp5) >= fabs(*old)) {
freq[lsp_index] = omega - step;
} else {
freq[lsp_index] = omega;
}
if ((*old) >= 0.0){
*old = (float)-1.0 * FLOAT_MAX;
} else {
*old = FLOAT_MAX;
}
omega = old_omega;
step_idx = 0;
step_idx = LSF_NUMBER_OF_STEPS;
} else {
if (step_idx == 0) {
old_omega = omega;
}
step_idx++;
omega -= steps[step_idx];
/* Go back one grid step */
step = steps[step_idx];
}
} else {
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/* increment omega until they are of different sign,
and we know there is at least one root between omega
and old_omega */
*old = hlp5;
omega += step;
}
}
}
for (i = 0; i<LPC_FILTERORDER; i++) {
freq[i] = freq[i] * TWO_PI;
}
}
/*----------------------------------------------------------------*
* conversion from lsf coefficients to lpc coefficients
*---------------------------------------------------------------*/
void lsf2a(
float *a_coef, /* (o) lpc coefficients */
float *freq /* (i) lsf coefficients */
){
int i, j;
float hlp;
float p[LPC_HALFORDER], q[LPC_HALFORDER];
float a[LPC_HALFORDER + 1], a1[LPC_HALFORDER],
a2[LPC_HALFORDER];
float b[LPC_HALFORDER + 1], b1[LPC_HALFORDER],
b2[LPC_HALFORDER];
for (i=0; i<LPC_FILTERORDER; i++) {
freq[i] = freq[i] * PI2;
}
/* Check input for ill-conditioned cases. This part is not
found in the TIA standard. It involves the following 2 IF
blocks. If "freq" is judged ill-conditioned, then we first
modify freq[0] and freq[LPC_HALFORDER-1] (normally
LPC_HALFORDER = 10 for LPC applications), then we adjust
the other "freq" values slightly */
if ((freq[0] <= 0.0) || (freq[LPC_FILTERORDER - 1] >= 0.5)){
if (freq[0] <= 0.0) {
freq[0] = (float)0.022;
}
if (freq[LPC_FILTERORDER - 1] >= 0.5) {
freq[LPC_FILTERORDER - 1] = (float)0.499;
}
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hlp = (freq[LPC_FILTERORDER - 1] - freq[0]) /
(float) (LPC_FILTERORDER - 1);
for (i=1; i<LPC_FILTERORDER; i++) {
freq[i] = freq[i - 1] + hlp;
}
}
memset(a1, 0, LPC_HALFORDER*sizeof(float));
memset(a2, 0, LPC_HALFORDER*sizeof(float));
memset(b1, 0, LPC_HALFORDER*sizeof(float));
memset(b2, 0, LPC_HALFORDER*sizeof(float));
memset(a, 0, (LPC_HALFORDER+1)*sizeof(float));
memset(b, 0, (LPC_HALFORDER+1)*sizeof(float));
/* p[i] and q[i] compute cos(2*pi*omega_{2j}) and
cos(2*pi*omega_{2j-1} in eqs. 4.2.2.2-1 and 4.2.2.2-2.
Note that for this code p[i] specifies the coefficients
used in .Q_A(z) while q[i] specifies the coefficients used
in .P_A(z) */
for (i=0; i<LPC_HALFORDER; i++) {
p[i] = (float)cos(TWO_PI * freq[2 * i]);
q[i] = (float)cos(TWO_PI * freq[2 * i + 1]);
}
a[0] = 0.25;
b[0] = 0.25;
for (i= 0; i<LPC_HALFORDER; i++) {
a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
a2[i] = a1[i];
a1[i] = a[i];
b2[i] = b1[i];
b1[i] = b[i];
}
for (j=0; j<LPC_FILTERORDER; j++) {
if (j == 0) {
a[0] = 0.25;
b[0] = -0.25;
} else {
a[0] = b[0] = 0.0;
}
for (i=0; i<LPC_HALFORDER; i++) {
a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
a2[i] = a1[i];
a1[i] = a[i];
b2[i] = b1[i];
b1[i] = b[i];
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}
a_coef[j + 1] = 2 * (a[LPC_HALFORDER] + b[LPC_HALFORDER]);
}
a_coef[0] = 1.0;
}
A.41 packing.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
packing.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __PACKING_H
#define __PACKING_H
void packsplit(
int *index, /* (i) the value to split */
int *firstpart, /* (o) the value specified by most
significant bits */
int *rest, /* (o) the value specified by least
significant bits */
int bitno_firstpart, /* (i) number of bits in most
significant part */
int bitno_total /* (i) number of bits in full range
of value */
);
void packcombine(
int *index, /* (i/o) the msb value in the
combined value out */
int rest, /* (i) the lsb value */
int bitno_rest /* (i) the number of bits in the
lsb part */
);
void dopack(
unsigned char **bitstream, /* (i/o) on entrance pointer to
place in bitstream to pack
new data, on exit pointer
to place in bitstream to
pack future data */
int index, /* (i) the value to pack */
int bitno, /* (i) the number of bits that the
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value will fit within */
int *pos /* (i/o) write position in the
current byte */
);
void unpack(
unsigned char **bitstream, /* (i/o) on entrance pointer to
place in bitstream to
unpack new data from, on
exit pointer to place in
bitstream to unpack future
data from */
int *index, /* (o) resulting value */
int bitno, /* (i) number of bits used to
represent the value */
int *pos /* (i/o) read position in the
current byte */
);
#endif
A.42 packing.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
packing.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <stdlib.h>
#include "iLBC_define.h"
#include "constants.h"
#include "helpfun.h"
#include "string.h"
/*----------------------------------------------------------------*
* splitting an integer into first most significant bits and
* remaining least significant bits
*---------------------------------------------------------------*/
void packsplit(
int *index, /* (i) the value to split */
int *firstpart, /* (o) the value specified by most
significant bits */
int *rest, /* (o) the value specified by least
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significant bits */
int bitno_firstpart, /* (i) number of bits in most
significant part */
int bitno_total /* (i) number of bits in full range
of value */
){
int bitno_rest = bitno_total-bitno_firstpart;
*firstpart = *index>>(bitno_rest);
*rest = *index-(*firstpart<<(bitno_rest));
}
/*----------------------------------------------------------------*
* combining a value corresponding to msb's with a value
* corresponding to lsb's
*---------------------------------------------------------------*/
void packcombine(
int *index, /* (i/o) the msb value in the
combined value out */
int rest, /* (i) the lsb value */
int bitno_rest /* (i) the number of bits in the
lsb part */
){
*index = *index<<bitno_rest;
*index += rest;
}
/*----------------------------------------------------------------*
* packing of bits into bitstream, i.e., vector of bytes
*---------------------------------------------------------------*/
void dopack(
unsigned char **bitstream, /* (i/o) on entrance pointer to
place in bitstream to pack
new data, on exit pointer
to place in bitstream to
pack future data */
int index, /* (i) the value to pack */
int bitno, /* (i) the number of bits that the
value will fit within */
int *pos /* (i/o) write position in the
current byte */
){
int posLeft;
/* Clear the bits before starting in a new byte */
if ((*pos)==0) {
**bitstream=0;
}
while (bitno>0) {
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/* Jump to the next byte if end of this byte is reached*/
if (*pos==8) {
*pos=0;
(*bitstream)++;
**bitstream=0;
}
posLeft=8-(*pos);
/* Insert index into the bitstream */
if (bitno <= posLeft) {
**bitstream |= (unsigned char)(index<<(posLeft-bitno));
*pos+=bitno;
bitno=0;
} else {
**bitstream |= (unsigned char)(index>>(bitno-posLeft));
*pos=8;
index-=((index>>(bitno-posLeft))<<(bitno-posLeft));
bitno-=posLeft;
}
}
}
/*----------------------------------------------------------------*
* unpacking of bits from bitstream, i.e., vector of bytes
*---------------------------------------------------------------*/
void unpack(
unsigned char **bitstream, /* (i/o) on entrance pointer to
place in bitstream to
unpack new data from, on
exit pointer to place in
bitstream to unpack future
data from */
int *index, /* (o) resulting value */
int bitno, /* (i) number of bits used to
represent the value */
int *pos /* (i/o) read position in the
current byte */
){
int BitsLeft;
*index=0;
while (bitno>0) {
/* move forward in bitstream when the end of the
byte is reached */
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if (*pos==8) {
*pos=0;
(*bitstream)++;
}
BitsLeft=8-(*pos);
/* Extract bits to index */
if (BitsLeft>=bitno) {
*index+=((((**bitstream)<<(*pos)) & 0xFF)>>(8-bitno));
*pos+=bitno;
bitno=0;
} else {
if ((8-bitno)>0) {
*index+=((((**bitstream)<<(*pos)) & 0xFF)>>
(8-bitno));
*pos=8;
} else {
*index+=(((int)(((**bitstream)<<(*pos)) & 0xFF))<<
(bitno-8));
*pos=8;
}
bitno-=BitsLeft;
}
}
}
A.43 StateConstructW.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
StateConstructW.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_STATECONSTRUCTW_H
#define __iLBC_STATECONSTRUCTW_H
void StateConstructW(
int idxForMax, /* (i) 6-bit index for the quantization of
max amplitude */
int *idxVec, /* (i) vector of quantization indexes */
float *syntDenum, /* (i) synthesis filter denumerator */
float *out, /* (o) the decoded state vector */
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int len /* (i) length of a state vector */
);
#endif
A.44 StateConstructW.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
StateConstructW.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <string.h>
#include "iLBC_define.h"
#include "constants.h"
#include "filter.h"
/*----------------------------------------------------------------*
* decoding of the start state
*---------------------------------------------------------------*/
void StateConstructW(
int idxForMax, /* (i) 6-bit index for the quantization of
max amplitude */
int *idxVec, /* (i) vector of quantization indexes */
float *syntDenum, /* (i) synthesis filter denumerator */
float *out, /* (o) the decoded state vector */
int len /* (i) length of a state vector */
){
float maxVal, tmpbuf[LPC_FILTERORDER+2*STATE_LEN], *tmp,
numerator[LPC_FILTERORDER+1];
float foutbuf[LPC_FILTERORDER+2*STATE_LEN], *fout;
int k,tmpi;
/* decoding of the maximum value */
maxVal = state_frgqTbl[idxForMax];
maxVal = (float)pow(10,maxVal)/(float)4.5;
/* initialization of buffers and coefficients */
memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
for (k=0; k<LPC_FILTERORDER; k++) {
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numerator[k]=syntDenum[LPC_FILTERORDER-k];
}
numerator[LPC_FILTERORDER]=syntDenum[0];
tmp = &tmpbuf[LPC_FILTERORDER];
fout = &foutbuf[LPC_FILTERORDER];
/* decoding of the sample values */
for (k=0; k<len; k++) {
tmpi = len-1-k;
/* maxVal = 1/scal */
tmp[k] = maxVal*state_sq3Tbl[idxVec[tmpi]];
}
/* circular convolution with all-pass filter */
memset(tmp+len, 0, len*sizeof(float));
ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
LPC_FILTERORDER, fout);
for (k=0;k<len;k++) {
out[k] = fout[len-1-k]+fout[2*len-1-k];
}
}
A.45 StateSearchW.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
StateSearchW.h
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_STATESEARCHW_H
#define __iLBC_STATESEARCHW_H
void AbsQuantW(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) Encoder instance */
float *in, /* (i) vector to encode */
float *syntDenum, /* (i) denominator of synthesis filter */
float *weightDenum, /* (i) denominator of weighting filter */
int *out, /* (o) vector of quantizer indexes */
int len, /* (i) length of vector to encode and
vector of quantizer indexes */
int state_first /* (i) position of start state in the
80 vec */
);
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void StateSearchW(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) Encoder instance */
float *residual,/* (i) target residual vector */
float *syntDenum, /* (i) lpc synthesis filter */
float *weightDenum, /* (i) weighting filter denuminator */
int *idxForMax, /* (o) quantizer index for maximum
amplitude */
int *idxVec, /* (o) vector of quantization indexes */
int len, /* (i) length of all vectors */
int state_first /* (i) position of start state in the
80 vec */
);
#endif
A.46 StateSearchW.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
StateSearchW.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include <math.h>
#include <string.h>
#include "iLBC_define.h"
#include "constants.h"
#include "filter.h"
#include "helpfun.h"
/*----------------------------------------------------------------*
* predictive noise shaping encoding of scaled start state
* (subrutine for StateSearchW)
*---------------------------------------------------------------*/
void AbsQuantW(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) Encoder instance */
float *in, /* (i) vector to encode */
float *syntDenum, /* (i) denominator of synthesis filter */
float *weightDenum, /* (i) denominator of weighting filter */
int *out, /* (o) vector of quantizer indexes */
int len, /* (i) length of vector to encode and
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vector of quantizer indexes */
int state_first /* (i) position of start state in the
80 vec */
){
float *syntOut;
float syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
float toQ, xq;
int n;
int index;
/* initialization of buffer for filtering */
memset(syntOutBuf, 0, LPC_FILTERORDER*sizeof(float));
/* initialization of pointer for filtering */
syntOut = &syntOutBuf[LPC_FILTERORDER];
/* synthesis and weighting filters on input */
if (state_first) {
AllPoleFilter (in, weightDenum, SUBL, LPC_FILTERORDER);
} else {
AllPoleFilter (in, weightDenum,
iLBCenc_inst->state_short_len-SUBL,
LPC_FILTERORDER);
}
/* encoding loop */
for (n=0; n<len; n++) {
/* time update of filter coefficients */
if ((state_first)&&(n==SUBL)){
syntDenum += (LPC_FILTERORDER+1);
weightDenum += (LPC_FILTERORDER+1);
/* synthesis and weighting filters on input */
AllPoleFilter (&in[n], weightDenum, len-n,
LPC_FILTERORDER);
} else if ((state_first==0)&&
(n==(iLBCenc_inst->state_short_len-SUBL))) {
syntDenum += (LPC_FILTERORDER+1);
weightDenum += (LPC_FILTERORDER+1);
/* synthesis and weighting filters on input */
AllPoleFilter (&in[n], weightDenum, len-n,
LPC_FILTERORDER);
}
/* prediction of synthesized and weighted input */
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syntOut[n] = 0.0;
AllPoleFilter (&syntOut[n], weightDenum, 1,
LPC_FILTERORDER);
/* quantization */
toQ = in[n]-syntOut[n];
sort_sq(&xq, &index, toQ, state_sq3Tbl, 8);
out[n]=index;
syntOut[n] = state_sq3Tbl[out[n]];
/* update of the prediction filter */
AllPoleFilter(&syntOut[n], weightDenum, 1,
LPC_FILTERORDER);
}
}
/*----------------------------------------------------------------*
* encoding of start state
*---------------------------------------------------------------*/
void StateSearchW(
iLBC_Enc_Inst_t *iLBCenc_inst,
/* (i) Encoder instance */
float *residual,/* (i) target residual vector */
float *syntDenum, /* (i) lpc synthesis filter */
float *weightDenum, /* (i) weighting filter denuminator */
int *idxForMax, /* (o) quantizer index for maximum
amplitude */
int *idxVec, /* (o) vector of quantization indexes */
int len, /* (i) length of all vectors */
int state_first /* (i) position of start state in the
80 vec */
){
float dtmp, maxVal;
float tmpbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS];
float *tmp, numerator[1+LPC_FILTERORDER];
float foutbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS], *fout;
int k;
float qmax, scal;
/* initialization of buffers and filter coefficients */
memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
for (k=0; k<LPC_FILTERORDER; k++) {
numerator[k]=syntDenum[LPC_FILTERORDER-k];
}
numerator[LPC_FILTERORDER]=syntDenum[0];
tmp = &tmpbuf[LPC_FILTERORDER];
fout = &foutbuf[LPC_FILTERORDER];
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/* circular convolution with the all-pass filter */
memcpy(tmp, residual, len*sizeof(float));
memset(tmp+len, 0, len*sizeof(float));
ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
LPC_FILTERORDER, fout);
for (k=0; k<len; k++) {
fout[k] += fout[k+len];
}
/* identification of the maximum amplitude value */
maxVal = fout[0];
for (k=1; k<len; k++) {
if (fout[k]*fout[k] > maxVal*maxVal){
maxVal = fout[k];
}
}
maxVal=(float)fabs(maxVal);
/* encoding of the maximum amplitude value */
if (maxVal < 10.0) {
maxVal = 10.0;
}
maxVal = (float)log10(maxVal);
sort_sq(&dtmp, idxForMax, maxVal, state_frgqTbl, 64);
/* decoding of the maximum amplitude representation value,
and corresponding scaling of start state */
maxVal=state_frgqTbl[*idxForMax];
qmax = (float)pow(10,maxVal);
scal = (float)(4.5)/qmax;
for (k=0; k<len; k++){
fout[k] *= scal;
}
/* predictive noise shaping encoding of scaled start state */
AbsQuantW(iLBCenc_inst, fout,syntDenum,
weightDenum,idxVec, len, state_first);
}
A.47 syntFilter.h
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
syntFilter.h
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Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#ifndef __iLBC_SYNTFILTER_H
#define __iLBC_SYNTFILTER_H
void syntFilter(
float *Out, /* (i/o) Signal to be filtered */
float *a, /* (i) LP parameters */
int len, /* (i) Length of signal */
float *mem /* (i/o) Filter state */
);
#endif
A.48 syntFilter.c
/******************************************************************
iLBC Speech Coder ANSI-C Source Code
syntFilter.c
Copyright (c) 2001-2003,
Global IP Sound AB.
All rights reserved.
******************************************************************/
#include "iLBC_define.h"
/*----------------------------------------------------------------*
* LP synthesis filter.
*---------------------------------------------------------------*/
void syntFilter(
float *Out, /* (i/o) Signal to be filtered */
float *a, /* (i) LP parameters */
int len, /* (i) Length of signal */
float *mem /* (i/o) Filter state */
){
int i, j;
float *po, *pi, *pa, *pm;
po=Out;
/* Filter first part using memory from past */
for (i=0; i<LPC_FILTERORDER; i++) {
pi=&Out[i-1];
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pa=&a[1];
pm=&mem[LPC_FILTERORDER-1];
for (j=1; j<=i; j++) {
*po-=(*pa++)*(*pi--);
}
for (j=i+1; j<LPC_FILTERORDER+1; j++) {
*po-=(*pa++)*(*pm--);
}
po++;
}
/* Filter last part where the state is entierly in
the output vector */
for (i=LPC_FILTERORDER; i<len; i++) {
pi=&Out[i-1];
pa=&a[1];
for (j=1; j<LPC_FILTERORDER+1; j++) {
*po-=(*pa++)*(*pi--);
}
po++;
}
/* Update state vector */
memcpy(mem, &Out[len-LPC_FILTERORDER],
LPC_FILTERORDER*sizeof(float));
}
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