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freeswitch/libs/libcodec2/src/quantise.c

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/*---------------------------------------------------------------------------*\
FILE........: quantise.c
AUTHOR......: David Rowe
DATE CREATED: 31/5/92
Quantisation functions for the sinusoidal coder.
\*---------------------------------------------------------------------------*/
/*
All rights reserved.
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License version 2.1, as
published by the Free Software Foundation. This program is
distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or
FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public
License for more details.
You should have received a copy of the GNU Lesser General Public License
along with this program; if not, see <http://www.gnu.org/licenses/>.
*/
#include <assert.h>
#include <ctype.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "defines.h"
#include "dump.h"
#include "quantise.h"
#include "lpc.h"
#include "lsp.h"
#include "kiss_fft.h"
#define LSP_DELTA1 0.01 /* grid spacing for LSP root searches */
/*---------------------------------------------------------------------------*\
FUNCTION HEADERS
\*---------------------------------------------------------------------------*/
float speech_to_uq_lsps(float lsp[], float ak[], float Sn[], float w[],
int order);
/*---------------------------------------------------------------------------*\
FUNCTIONS
\*---------------------------------------------------------------------------*/
int lsp_bits(int i) {
return lsp_cb[i].log2m;
}
int lspd_bits(int i) {
return lsp_cbd[i].log2m;
}
int lspdt_bits(int i) {
return lsp_cbdt[i].log2m;
}
int lsp_pred_vq_bits(int i) {
return lsp_cbjvm[i].log2m;
}
/*---------------------------------------------------------------------------*\
quantise_init
Loads the entire LSP quantiser comprised of several vector quantisers
(codebooks).
\*---------------------------------------------------------------------------*/
void quantise_init()
{
}
/*---------------------------------------------------------------------------*\
quantise
Quantises vec by choosing the nearest vector in codebook cb, and
returns the vector index. The squared error of the quantised vector
is added to se.
\*---------------------------------------------------------------------------*/
long quantise(const float * cb, float vec[], float w[], int k, int m, float *se)
/* float cb[][K]; current VQ codebook */
/* float vec[]; vector to quantise */
/* float w[]; weighting vector */
/* int k; dimension of vectors */
/* int m; size of codebook */
/* float *se; accumulated squared error */
{
float e; /* current error */
long besti; /* best index so far */
float beste; /* best error so far */
long j;
int i;
float diff;
besti = 0;
beste = 1E32;
for(j=0; j<m; j++) {
e = 0.0;
for(i=0; i<k; i++) {
diff = cb[j*k+i]-vec[i];
e += pow(diff*w[i],2.0);
}
if (e < beste) {
beste = e;
besti = j;
}
}
*se += beste;
return(besti);
}
/*---------------------------------------------------------------------------*\
encode_lspds_scalar()
Scalar/VQ LSP difference quantiser.
\*---------------------------------------------------------------------------*/
void encode_lspds_scalar(
int indexes[],
float lsp[],
int order
)
{
int i,k,m;
float lsp_hz[LPC_MAX];
float lsp__hz[LPC_MAX];
float dlsp[LPC_MAX];
float dlsp_[LPC_MAX];
float wt[LPC_MAX];
const float *cb;
float se;
assert(order == LPC_ORD);
for(i=0; i<order; i++) {
wt[i] = 1.0;
}
/* convert from radians to Hz so we can use human readable
frequencies */
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
//printf("\n");
wt[0] = 1.0;
for(i=0; i<order; i++) {
/* find difference from previous qunatised lsp */
if (i)
dlsp[i] = lsp_hz[i] - lsp__hz[i-1];
else
dlsp[0] = lsp_hz[0];
k = lsp_cbd[i].k;
m = lsp_cbd[i].m;
cb = lsp_cbd[i].cb;
indexes[i] = quantise(cb, &dlsp[i], wt, k, m, &se);
dlsp_[i] = cb[indexes[i]*k];
if (i)
lsp__hz[i] = lsp__hz[i-1] + dlsp_[i];
else
lsp__hz[0] = dlsp_[0];
//printf("%d lsp %3.2f dlsp %3.2f dlsp_ %3.2f lsp_ %3.2f\n", i, lsp_hz[i], dlsp[i], dlsp_[i], lsp__hz[i]);
}
}
void decode_lspds_scalar(
float lsp_[],
int indexes[],
int order
)
{
int i,k;
float lsp__hz[LPC_MAX];
float dlsp_[LPC_MAX];
const float *cb;
assert(order == LPC_ORD);
for(i=0; i<order; i++) {
k = lsp_cbd[i].k;
cb = lsp_cbd[i].cb;
dlsp_[i] = cb[indexes[i]*k];
if (i)
lsp__hz[i] = lsp__hz[i-1] + dlsp_[i];
else
lsp__hz[0] = dlsp_[0];
lsp_[i] = (PI/4000.0)*lsp__hz[i];
//printf("%d dlsp_ %3.2f lsp_ %3.2f\n", i, dlsp_[i], lsp__hz[i]);
}
}
/*---------------------------------------------------------------------------*\
lspvq_quantise
Vector LSP quantiser.
\*---------------------------------------------------------------------------*/
void lspvq_quantise(
float lsp[],
float lsp_[],
int order
)
{
int i,k,m,ncb, nlsp;
float wt[LPC_ORD], lsp_hz[LPC_ORD];
const float *cb;
float se;
int index;
for(i=0; i<LPC_ORD; i++) {
wt[i] = 1.0;
lsp_hz[i] = 4000.0*lsp[i]/PI;
}
/* scalar quantise LSPs 1,2,3,4 */
/* simple uniform scalar quantisers */
for(i=0; i<4; i++) {
k = lsp_cb[i].k;
m = lsp_cb[i].m;
cb = lsp_cb[i].cb;
index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
lsp_[i] = cb[index*k]*PI/4000.0;
}
//#define WGHT
#ifdef WGHT
for(i=4; i<9; i++) {
wt[i] = 1.0/(lsp[i]-lsp[i-1]) + 1.0/(lsp[i+1]-lsp[i]);
//printf("wt[%d] = %f\n", i, wt[i]);
}
wt[9] = 1.0/(lsp[i]-lsp[i-1]);
#endif
/* VQ LSPs 5,6,7,8,9,10 */
ncb = 4;
nlsp = 4;
k = lsp_cbjnd[ncb].k;
m = lsp_cbjnd[ncb].m;
cb = lsp_cbjnd[ncb].cb;
index = quantise(cb, &lsp_hz[nlsp], &wt[nlsp], k, m, &se);
for(i=4; i<LPC_ORD; i++) {
lsp_[i] = cb[index*k+i-4]*(PI/4000.0);
//printf("%4.f (%4.f) ", lsp_hz[i], cb[index*k+i-4]);
}
}
/*---------------------------------------------------------------------------*\
lspjnd_quantise
Experimental JND LSP quantiser.
\*---------------------------------------------------------------------------*/
void lspjnd_quantise(float lsps[], float lsps_[], int order)
{
int i,k,m;
float wt[LPC_ORD], lsps_hz[LPC_ORD];
const float *cb;
float se = 0.0;
int index;
for(i=0; i<LPC_ORD; i++) {
wt[i] = 1.0;
}
/* convert to Hz */
for(i=0; i<LPC_ORD; i++) {
lsps_hz[i] = lsps[i]*(4000.0/PI);
lsps_[i] = lsps[i];
}
/* simple uniform scalar quantisers */
for(i=0; i<4; i++) {
k = lsp_cbjnd[i].k;
m = lsp_cbjnd[i].m;
cb = lsp_cbjnd[i].cb;
index = quantise(cb, &lsps_hz[i], wt, k, m, &se);
lsps_[i] = cb[index*k]*(PI/4000.0);
}
/* VQ LSPs 5,6,7,8,9,10 */
k = lsp_cbjnd[4].k;
m = lsp_cbjnd[4].m;
cb = lsp_cbjnd[4].cb;
index = quantise(cb, &lsps_hz[4], &wt[4], k, m, &se);
//printf("k = %d m = %d c[0] %f cb[k] %f\n", k,m,cb[0],cb[k]);
//printf("index = %4d: ", index);
for(i=4; i<LPC_ORD; i++) {
lsps_[i] = cb[index*k+i-4]*(PI/4000.0);
//printf("%4.f (%4.f) ", lsps_hz[i], cb[index*k+i-4]);
}
//printf("\n");
}
void compute_weights(const float *x, float *w, int ndim);
/*---------------------------------------------------------------------------*\
lspdt_quantise
LSP difference in time quantiser. Split VQ, encoding LSPs 1-4 with
one VQ, and LSPs 5-10 with a second. Update of previous lsp memory
is done outside of this function to handle dT between 10 or 20ms
frames.
mode action
------------------
LSPDT_ALL VQ LSPs 1-4 and 5-10
LSPDT_LOW Just VQ LSPs 1-4, for LSPs 5-10 just copy previous
LSPDT_HIGH Just VQ LSPs 5-10, for LSPs 1-4 just copy previous
\*---------------------------------------------------------------------------*/
void lspdt_quantise(float lsps[], float lsps_[], float lsps__prev[], int mode)
{
int i;
float wt[LPC_ORD];
float lsps_dt[LPC_ORD];
#ifdef TRY_LSPDT_VQ
int k,m;
int index;
const float *cb;
float se = 0.0;
#endif // TRY_LSPDT_VQ
//compute_weights(lsps, wt, LPC_ORD);
for(i=0; i<LPC_ORD; i++) {
wt[i] = 1.0;
}
//compute_weights(lsps, wt, LPC_ORD );
for(i=0; i<LPC_ORD; i++) {
lsps_dt[i] = lsps[i] - lsps__prev[i];
lsps_[i] = lsps__prev[i];
}
//#define TRY_LSPDT_VQ
#ifdef TRY_LSPDT_VQ
/* this actually improves speech a bit, but 40ms updates works surprsingly well.... */
k = lsp_cbdt[0].k;
m = lsp_cbdt[0].m;
cb = lsp_cbdt[0].cb;
index = quantise(cb, lsps_dt, wt, k, m, &se);
for(i=0; i<LPC_ORD; i++) {
lsps_[i] += cb[index*k + i];
}
#endif
}
#define MIN(a,b) ((a)<(b)?(a):(b))
#define MAX_ENTRIES 16384
void compute_weights(const float *x, float *w, int ndim)
{
int i;
w[0] = MIN(x[0], x[1]-x[0]);
for (i=1;i<ndim-1;i++)
w[i] = MIN(x[i]-x[i-1], x[i+1]-x[i]);
w[ndim-1] = MIN(x[ndim-1]-x[ndim-2], PI-x[ndim-1]);
for (i=0;i<ndim;i++)
w[i] = 1./(.01+w[i]);
//w[0]*=3;
//w[1]*=2;
}
/* LSP weight calculation ported from m-file function kindly submitted
by Anssi, OH3GDD */
void compute_weights_anssi_mode2(const float *x, float *w, int ndim)
{
int i;
float d[LPC_ORD];
assert(ndim == LPC_ORD);
for(i=0; i<LPC_ORD; i++)
d[i] = 1.0;
d[0] = x[1];
for (i=1; i<LPC_ORD-1; i++)
d[i] = x[i+1] - x[i-1];
d[LPC_ORD-1] = PI - x[8];
for (i=0; i<LPC_ORD; i++) {
if (x[i]<((400.0/4000.0)*PI))
w[i]=5.0/(0.01+d[i]);
else if (x[i]<((700.0/4000.0)*PI))
w[i]=4.0/(0.01+d[i]);
else if (x[i]<((1200.0/4000.0)*PI))
w[i]=3.0/(0.01+d[i]);
else if (x[i]<((2000.0/4000.0)*PI))
w[i]=2.0/(0.01+d[i]);
else
w[i]=1.0/(0.01+d[i]);
w[i]=pow(w[i]+0.3, 0.66);
}
}
int find_nearest(const float *codebook, int nb_entries, float *x, int ndim)
{
int i, j;
float min_dist = 1e15;
int nearest = 0;
for (i=0;i<nb_entries;i++)
{
float dist=0;
for (j=0;j<ndim;j++)
dist += (x[j]-codebook[i*ndim+j])*(x[j]-codebook[i*ndim+j]);
if (dist<min_dist)
{
min_dist = dist;
nearest = i;
}
}
return nearest;
}
int find_nearest_weighted(const float *codebook, int nb_entries, float *x, const float *w, int ndim)
{
int i, j;
float min_dist = 1e15;
int nearest = 0;
for (i=0;i<nb_entries;i++)
{
float dist=0;
for (j=0;j<ndim;j++)
dist += w[j]*(x[j]-codebook[i*ndim+j])*(x[j]-codebook[i*ndim+j]);
if (dist<min_dist)
{
min_dist = dist;
nearest = i;
}
}
return nearest;
}
void lspjvm_quantise(float *x, float *xq, int ndim)
{
int i, n1, n2, n3;
float err[LPC_ORD], err2[LPC_ORD], err3[LPC_ORD];
float w[LPC_ORD], w2[LPC_ORD], w3[LPC_ORD];
const float *codebook1 = lsp_cbjvm[0].cb;
const float *codebook2 = lsp_cbjvm[1].cb;
const float *codebook3 = lsp_cbjvm[2].cb;
w[0] = MIN(x[0], x[1]-x[0]);
for (i=1;i<ndim-1;i++)
w[i] = MIN(x[i]-x[i-1], x[i+1]-x[i]);
w[ndim-1] = MIN(x[ndim-1]-x[ndim-2], PI-x[ndim-1]);
compute_weights(x, w, ndim);
n1 = find_nearest(codebook1, lsp_cbjvm[0].m, x, ndim);
for (i=0;i<ndim;i++)
{
xq[i] = codebook1[ndim*n1+i];
err[i] = x[i] - xq[i];
}
for (i=0;i<ndim/2;i++)
{
err2[i] = err[2*i];
err3[i] = err[2*i+1];
w2[i] = w[2*i];
w3[i] = w[2*i+1];
}
n2 = find_nearest_weighted(codebook2, lsp_cbjvm[1].m, err2, w2, ndim/2);
n3 = find_nearest_weighted(codebook3, lsp_cbjvm[2].m, err3, w3, ndim/2);
for (i=0;i<ndim/2;i++)
{
xq[2*i] += codebook2[ndim*n2/2+i];
xq[2*i+1] += codebook3[ndim*n3/2+i];
}
}
#define MBEST_STAGES 4
struct MBEST_LIST {
int index[MBEST_STAGES]; /* index of each stage that lead us to this error */
float error;
};
struct MBEST {
int entries; /* number of entries in mbest list */
struct MBEST_LIST *list;
};
static struct MBEST *mbest_create(int entries) {
int i,j;
struct MBEST *mbest;
assert(entries > 0);
mbest = (struct MBEST *)malloc(sizeof(struct MBEST));
assert(mbest != NULL);
mbest->entries = entries;
mbest->list = (struct MBEST_LIST *)malloc(entries*sizeof(struct MBEST_LIST));
assert(mbest->list != NULL);
for(i=0; i<mbest->entries; i++) {
for(j=0; j<MBEST_STAGES; j++)
mbest->list[i].index[j] = 0;
mbest->list[i].error = 1E32;
}
return mbest;
}
static void mbest_destroy(struct MBEST *mbest) {
assert(mbest != NULL);
free(mbest->list);
free(mbest);
}
/*---------------------------------------------------------------------------*\
mbest_insert
Insert the results of a vector to codebook entry comparison. The
list is ordered in order or error, so those entries with the
smallest error will be first on the list.
\*---------------------------------------------------------------------------*/
static void mbest_insert(struct MBEST *mbest, int index[], float error) {
int i, j, found;
struct MBEST_LIST *list = mbest->list;
int entries = mbest->entries;
found = 0;
for(i=0; i<entries && !found; i++)
if (error < list[i].error) {
found = 1;
for(j=entries-1; j>i; j--)
list[j] = list[j-1];
for(j=0; j<MBEST_STAGES; j++)
list[i].index[j] = index[j];
list[i].error = error;
}
}
static void mbest_print(char title[], struct MBEST *mbest) {
int i,j;
printf("%s\n", title);
for(i=0; i<mbest->entries; i++) {
for(j=0; j<MBEST_STAGES; j++)
printf(" %4d ", mbest->list[i].index[j]);
printf(" %f\n", mbest->list[i].error);
}
}
/*---------------------------------------------------------------------------*\
mbest_search
Searches vec[] to a codebbook of vectors, and maintains a list of the mbest
closest matches.
\*---------------------------------------------------------------------------*/
static void mbest_search(
const float *cb, /* VQ codebook to search */
float vec[], /* target vector */
float w[], /* weighting vector */
int k, /* dimension of vector */
int m, /* number on entries in codebook */
struct MBEST *mbest, /* list of closest matches */
int index[] /* indexes that lead us here */
)
{
float e;
int i,j;
float diff;
for(j=0; j<m; j++) {
e = 0.0;
for(i=0; i<k; i++) {
diff = cb[j*k+i]-vec[i];
e += pow(diff*w[i],2.0);
}
index[0] = j;
mbest_insert(mbest, index, e);
}
}
/* 3 stage VQ LSP quantiser. Design and guidance kindly submitted by Anssi, OH3GDD */
void lspanssi_quantise(float *x, float *xq, int ndim, int mbest_entries)
{
int i, j, n1, n2, n3, n4;
float w[LPC_ORD];
const float *codebook1 = lsp_cbvqanssi[0].cb;
const float *codebook2 = lsp_cbvqanssi[1].cb;
const float *codebook3 = lsp_cbvqanssi[2].cb;
const float *codebook4 = lsp_cbvqanssi[3].cb;
struct MBEST *mbest_stage1, *mbest_stage2, *mbest_stage3, *mbest_stage4;
float target[LPC_ORD];
int index[MBEST_STAGES];
mbest_stage1 = mbest_create(mbest_entries);
mbest_stage2 = mbest_create(mbest_entries);
mbest_stage3 = mbest_create(mbest_entries);
mbest_stage4 = mbest_create(mbest_entries);
for(i=0; i<MBEST_STAGES; i++)
index[i] = 0;
compute_weights_anssi_mode2(x, w, ndim);
#ifdef DUMP
dump_weights(w, ndim);
#endif
/* Stage 1 */
mbest_search(codebook1, x, w, ndim, lsp_cbvqanssi[0].m, mbest_stage1, index);
mbest_print("Stage 1:", mbest_stage1);
/* Stage 2 */
for (j=0; j<mbest_entries; j++) {
index[1] = n1 = mbest_stage1->list[j].index[0];
for(i=0; i<ndim; i++)
target[i] = x[i] - codebook1[ndim*n1+i];
mbest_search(codebook2, target, w, ndim, lsp_cbvqanssi[1].m, mbest_stage2, index);
}
mbest_print("Stage 2:", mbest_stage2);
/* Stage 3 */
for (j=0; j<mbest_entries; j++) {
index[2] = n1 = mbest_stage2->list[j].index[1];
index[1] = n2 = mbest_stage2->list[j].index[0];
for(i=0; i<ndim; i++)
target[i] = x[i] - codebook1[ndim*n1+i] - codebook2[ndim*n2+i];
mbest_search(codebook3, target, w, ndim, lsp_cbvqanssi[2].m, mbest_stage3, index);
}
mbest_print("Stage 3:", mbest_stage3);
/* Stage 4 */
for (j=0; j<mbest_entries; j++) {
index[3] = n1 = mbest_stage3->list[j].index[2];
index[2] = n2 = mbest_stage3->list[j].index[1];
index[1] = n3 = mbest_stage3->list[j].index[0];
for(i=0; i<ndim; i++)
target[i] = x[i] - codebook1[ndim*n1+i] - codebook2[ndim*n2+i] - codebook3[ndim*n3+i];
mbest_search(codebook4, target, w, ndim, lsp_cbvqanssi[3].m, mbest_stage4, index);
}
mbest_print("Stage 4:", mbest_stage4);
n1 = mbest_stage4->list[0].index[3];
n2 = mbest_stage4->list[0].index[2];
n3 = mbest_stage4->list[0].index[1];
n4 = mbest_stage4->list[0].index[0];
for (i=0;i<ndim;i++)
xq[i] = codebook1[ndim*n1+i] + codebook2[ndim*n2+i] + codebook3[ndim*n3+i] + codebook4[ndim*n4+i];
mbest_destroy(mbest_stage1);
mbest_destroy(mbest_stage2);
mbest_destroy(mbest_stage3);
mbest_destroy(mbest_stage4);
}
int check_lsp_order(float lsp[], int lpc_order)
{
int i;
float tmp;
int swaps = 0;
for(i=1; i<lpc_order; i++)
if (lsp[i] < lsp[i-1]) {
//printf("swap %d\n",i);
swaps++;
tmp = lsp[i-1];
lsp[i-1] = lsp[i]-0.05;
lsp[i] = tmp+0.05;
}
return swaps;
}
void force_min_lsp_dist(float lsp[], int lpc_order)
{
int i;
for(i=1; i<lpc_order; i++)
if ((lsp[i]-lsp[i-1]) < 0.01) {
lsp[i] += 0.01;
}
}
#ifdef NOT_USED
/*---------------------------------------------------------------------------*\
lpc_model_amplitudes
Derive a LPC model for amplitude samples then estimate amplitude samples
from this model with optional LSP quantisation.
Returns the spectral distortion for this frame.
\*---------------------------------------------------------------------------*/
float lpc_model_amplitudes(
float Sn[], /* Input frame of speech samples */
float w[],
MODEL *model, /* sinusoidal model parameters */
int order, /* LPC model order */
int lsp_quant, /* optional LSP quantisation if non-zero */
float ak[] /* output aks */
)
{
float Wn[M];
float R[LPC_MAX+1];
float E;
int i,j;
float snr;
float lsp[LPC_MAX];
float lsp_hz[LPC_MAX];
float lsp_[LPC_MAX];
int roots; /* number of LSP roots found */
float wt[LPC_MAX];
for(i=0; i<M; i++)
Wn[i] = Sn[i]*w[i];
autocorrelate(Wn,R,M,order);
levinson_durbin(R,ak,order);
E = 0.0;
for(i=0; i<=order; i++)
E += ak[i]*R[i];
for(i=0; i<order; i++)
wt[i] = 1.0;
if (lsp_quant) {
roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
if (roots != order)
printf("LSP roots not found\n");
/* convert from radians to Hz to make quantisers more
human readable */
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
#ifdef NOT_NOW
/* simple uniform scalar quantisers */
for(i=0; i<10; i++) {
k = lsp_cb[i].k;
m = lsp_cb[i].m;
cb = lsp_cb[i].cb;
index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
lsp_hz[i] = cb[index*k];
}
#endif
/* experiment: simulating uniform quantisation error
for(i=0; i<order; i++)
lsp[i] += PI*(12.5/4000.0)*(1.0 - 2.0*(float)rand()/RAND_MAX);
*/
for(i=0; i<order; i++)
lsp[i] = (PI/4000.0)*lsp_hz[i];
/* Bandwidth Expansion (BW). Prevents any two LSPs getting too
close together after quantisation. We know from experiment
that LSP quantisation errors < 12.5Hz (25Hz step size) are
inaudible so we use that as the minimum LSP separation.
*/
for(i=1; i<5; i++) {
if (lsp[i] - lsp[i-1] < PI*(12.5/4000.0))
lsp[i] = lsp[i-1] + PI*(12.5/4000.0);
}
/* as quantiser gaps increased, larger BW expansion was required
to prevent twinkly noises */
for(i=5; i<8; i++) {
if (lsp[i] - lsp[i-1] < PI*(25.0/4000.0))
lsp[i] = lsp[i-1] + PI*(25.0/4000.0);
}
for(i=8; i<order; i++) {
if (lsp[i] - lsp[i-1] < PI*(75.0/4000.0))
lsp[i] = lsp[i-1] + PI*(75.0/4000.0);
}
for(j=0; j<order; j++)
lsp_[j] = lsp[j];
lsp_to_lpc(lsp_, ak, order);
#ifdef DUMP
dump_lsp(lsp);
#endif
}
#ifdef DUMP
dump_E(E);
#endif
#ifdef SIM_QUANT
/* simulated LPC energy quantisation */
{
float e = 10.0*log10(E);
e += 2.0*(1.0 - 2.0*(float)rand()/RAND_MAX);
E = pow(10.0,e/10.0);
}
#endif
aks_to_M2(ak,order,model,E,&snr, 1, 0, 1); /* {ak} -> {Am} LPC decode */
return snr;
}
#endif
/*---------------------------------------------------------------------------*\
lpc_post_filter()
Applies a post filter to the LPC synthesis filter power spectrum
Pw, which supresses the inter-formant energy.
The algorithm is from p267 (Section 8.6) of "Digital Speech",
edited by A.M. Kondoz, 1994 published by Wiley and Sons. Chapter 8
of this text is on the MBE vocoder, and this is a freq domain
adaptation of post filtering commonly used in CELP.
I used the Octave simulation lpcpf.m to get an understaing of the
algorithm.
Requires two more FFTs which is significantly more MIPs. However
it should be possible to implement this more efficiently in the
time domain. Just not sure how to handle relative time delays
between the synthesis stage and updating these coeffs. A smaller
FFT size might also be accetable to save CPU.
TODO:
[ ] sync var names between Octave and C version
[ ] doc gain normalisation
[ ] I think the first FFT is not rqd as we do the same
thing in aks_to_M2().
\*---------------------------------------------------------------------------*/
void lpc_post_filter(kiss_fft_cfg fft_fwd_cfg, MODEL *model, COMP Pw[], float ak[],
int order, int dump, float beta, float gamma, int bass_boost)
{
int i;
COMP x[FFT_ENC]; /* input to FFTs */
COMP Aw[FFT_ENC]; /* LPC analysis filter spectrum */
COMP Ww[FFT_ENC]; /* weighting spectrum */
float Rw[FFT_ENC]; /* R = WA */
float e_before, e_after, gain;
float Pfw[FFT_ENC]; /* Post filter mag spectrum */
float max_Rw, min_Rw;
float range, thresh, r, w;
int m, bin;
/* Determine LPC inverse filter spectrum 1/A(exp(jw)) -----------*/
/* we actually want the synthesis filter A(exp(jw)) but the
inverse (analysis) filter is easier to find as it's FIR, we
just use the inverse of 1/A to get the synthesis filter
A(exp(jw)) */
for(i=0; i<FFT_ENC; i++) {
x[i].real = 0.0;
x[i].imag = 0.0;
}
for(i=0; i<=order; i++)
x[i].real = ak[i];
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)x, (kiss_fft_cpx *)Aw);
for(i=0; i<FFT_ENC/2; i++) {
Aw[i].real = 1.0/sqrt(Aw[i].real*Aw[i].real + Aw[i].imag*Aw[i].imag);
}
/* Determine weighting filter spectrum W(exp(jw)) ---------------*/
for(i=0; i<FFT_ENC; i++) {
x[i].real = 0.0;
x[i].imag = 0.0;
}
for(i=0; i<=order; i++)
x[i].real = ak[i] * pow(gamma, (float)i);
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)x, (kiss_fft_cpx *)Ww);
for(i=0; i<FFT_ENC/2; i++) {
Ww[i].real = sqrt(Ww[i].real*Ww[i].real + Ww[i].imag*Ww[i].imag);
}
/* Determined combined filter R = WA ---------------------------*/
max_Rw = 0.0; min_Rw = 1E32;
for(i=0; i<FFT_ENC/2; i++) {
Rw[i] = Ww[i].real * Aw[i].real;
if (Rw[i] > max_Rw)
max_Rw = Rw[i];
if (Rw[i] < min_Rw)
min_Rw = Rw[i];
}
#ifdef DUMP
if (dump)
dump_Rw(Rw);
#endif
/* create post filter mag spectrum and apply ------------------*/
/* measure energy before post filtering */
e_before = 1E-4;
for(i=0; i<FFT_ENC/2; i++)
e_before += Pw[i].real;
/* apply post filter and measure energy */
#ifdef DUMP
if (dump)
dump_Pwb(Pw);
#endif
e_after = 1E-4;
for(i=0; i<FFT_ENC/2; i++) {
Pfw[i] = pow(Rw[i], beta);
Pw[i].real *= Pfw[i] * Pfw[i];
e_after += Pw[i].real;
}
gain = e_before/e_after;
/* apply gain factor to normalise energy */
for(i=0; i<FFT_ENC/2; i++) {
Pw[i].real *= gain;
}
if (bass_boost) {
/* add 3dB to first 1 kHz to account for LP effect of PF */
for(i=0; i<FFT_ENC/8; i++) {
Pw[i].real *= 1.4*1.4;
}
}
}
/*---------------------------------------------------------------------------*\
aks_to_M2()
Transforms the linear prediction coefficients to spectral amplitude
samples. This function determines A(m) from the average energy per
band using an FFT.
\*---------------------------------------------------------------------------*/
void aks_to_M2(
kiss_fft_cfg fft_fwd_cfg,
float ak[], /* LPC's */
int order,
MODEL *model, /* sinusoidal model parameters for this frame */
float E, /* energy term */
float *snr, /* signal to noise ratio for this frame in dB */
int dump, /* true to dump sample to dump file */
int sim_pf, /* true to simulate a post filter */
int pf, /* true to LPC post filter */
int bass_boost, /* enable LPC filter 0-1khz 3dB boost */
float beta,
float gamma /* LPC post filter parameters */
)
{
COMP pw[FFT_ENC]; /* input to FFT for power spectrum */
COMP Pw[FFT_ENC]; /* output power spectrum */
int i,m; /* loop variables */
int am,bm; /* limits of current band */
float r; /* no. rads/bin */
float Em; /* energy in band */
float Am; /* spectral amplitude sample */
float signal, noise;
r = TWO_PI/(FFT_ENC);
/* Determine DFT of A(exp(jw)) --------------------------------------------*/
for(i=0; i<FFT_ENC; i++) {
pw[i].real = 0.0;
pw[i].imag = 0.0;
}
for(i=0; i<=order; i++)
pw[i].real = ak[i];
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)pw, (kiss_fft_cpx *)Pw);
/* Determine power spectrum P(w) = E/(A(exp(jw))^2 ------------------------*/
for(i=0; i<FFT_ENC/2; i++)
Pw[i].real = E/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
if (pf)
lpc_post_filter(fft_fwd_cfg, model, Pw, ak, order, dump, beta, gamma, bass_boost);
#ifdef DUMP
if (dump)
dump_Pw(Pw);
#endif
/* Determine magnitudes from P(w) ----------------------------------------*/
/* when used just by decoder {A} might be all zeroes so init signal
and noise to prevent log(0) errors */
signal = 1E-30; noise = 1E-32;
for(m=1; m<=model->L; m++) {
am = floor((m - 0.5)*model->Wo/r + 0.5);
bm = floor((m + 0.5)*model->Wo/r + 0.5);
Em = 0.0;
for(i=am; i<bm; i++)
Em += Pw[i].real;
Am = sqrt(Em);
signal += pow(model->A[m],2.0);
noise += pow(model->A[m] - Am,2.0);
/* This code significantly improves perf of LPC model, in
particular when combined with phase0. The LPC spectrum tends
to track just under the peaks of the spectral envelope, and
just above nulls. This algorithm does the reverse to
compensate - raising the amplitudes of spectral peaks, while
attenuating the null. This enhances the formants, and
supresses the energy between formants. */
if (sim_pf) {
if (Am > model->A[m])
Am *= 0.7;
if (Am < model->A[m])
Am *= 1.4;
}
model->A[m] = Am;
}
*snr = 10.0*log10(signal/noise);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_Wo()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Encodes Wo using a WO_LEVELS quantiser.
\*---------------------------------------------------------------------------*/
int encode_Wo(float Wo)
{
int index;
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
float norm;
norm = (Wo - Wo_min)/(Wo_max - Wo_min);
index = floor(WO_LEVELS * norm + 0.5);
if (index < 0 ) index = 0;
if (index > (WO_LEVELS-1)) index = WO_LEVELS-1;
return index;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_Wo()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Decodes Wo using a WO_LEVELS quantiser.
\*---------------------------------------------------------------------------*/
float decode_Wo(int index)
{
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
float step;
float Wo;
step = (Wo_max - Wo_min)/WO_LEVELS;
Wo = Wo_min + step*(index);
return Wo;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_Wo_dt()
AUTHOR......: David Rowe
DATE CREATED: 6 Nov 2011
Encodes Wo difference from last frame.
\*---------------------------------------------------------------------------*/
int encode_Wo_dt(float Wo, float prev_Wo)
{
int index, mask, max_index, min_index;
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
float norm;
norm = (Wo - prev_Wo)/(Wo_max - Wo_min);
index = floor(WO_LEVELS * norm + 0.5);
//printf("ENC index: %d ", index);
/* hard limit */
max_index = (1 << (WO_DT_BITS-1)) - 1;
min_index = - (max_index+1);
if (index > max_index) index = max_index;
if (index < min_index) index = min_index;
//printf("max_index: %d min_index: %d hard index: %d ",
// max_index, min_index, index);
/* mask so that only LSB WO_DT_BITS remain, bit WO_DT_BITS is the sign bit */
mask = ((1 << WO_DT_BITS) - 1);
index &= mask;
//printf("mask: 0x%x index: 0x%x\n", mask, index);
return index;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_Wo_dt()
AUTHOR......: David Rowe
DATE CREATED: 6 Nov 2011
Decodes Wo using WO_DT_BITS difference from last frame.
\*---------------------------------------------------------------------------*/
float decode_Wo_dt(int index, float prev_Wo)
{
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
float step;
float Wo;
int mask;
/* sign extend index */
//printf("DEC index: %d ");
if (index & (1 << (WO_DT_BITS-1))) {
mask = ~((1 << WO_DT_BITS) - 1);
index |= mask;
}
//printf("DEC mask: 0x%x index: %d \n", mask, index);
step = (Wo_max - Wo_min)/WO_LEVELS;
Wo = prev_Wo + step*(index);
/* bit errors can make us go out of range leading to all sorts of
probs like seg faults */
if (Wo > Wo_max) Wo = Wo_max;
if (Wo < Wo_min) Wo = Wo_min;
return Wo;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: speech_to_uq_lsps()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Analyse a windowed frame of time domain speech to determine LPCs
which are the converted to LSPs for quantisation and transmission
over the channel.
\*---------------------------------------------------------------------------*/
float speech_to_uq_lsps(float lsp[],
float ak[],
float Sn[],
float w[],
int order
)
{
int i, roots;
float Wn[M];
float R[LPC_MAX+1];
float e, E;
e = 0.0;
for(i=0; i<M; i++) {
Wn[i] = Sn[i]*w[i];
e += Wn[i]*Wn[i];
}
/* trap 0 energy case as LPC analysis will fail */
if (e == 0.0) {
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
return 0.0;
}
autocorrelate(Wn, R, M, order);
levinson_durbin(R, ak, order);
E = 0.0;
for(i=0; i<=order; i++)
E += ak[i]*R[i];
/* 15 Hz BW expansion as I can't hear the difference and it may help
help occasional fails in the LSP root finding. Important to do this
after energy calculation to avoid -ve energy values.
*/
for(i=0; i<=order; i++)
ak[i] *= pow(0.994,(float)i);
roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
if (roots != order) {
/* if root finding fails use some benign LSP values instead */
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
}
return E;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_lsps_scalar()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Thirty-six bit sclar LSP quantiser. From a vector of unquantised
(floating point) LSPs finds the quantised LSP indexes.
\*---------------------------------------------------------------------------*/
void encode_lsps_scalar(int indexes[], float lsp[], int order)
{
int i,k,m;
float wt[1];
float lsp_hz[LPC_MAX];
const float * cb;
float se;
/* convert from radians to Hz so we can use human readable
frequencies */
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
/* scalar quantisers */
wt[0] = 1.0;
for(i=0; i<order; i++) {
k = lsp_cb[i].k;
m = lsp_cb[i].m;
cb = lsp_cb[i].cb;
indexes[i] = quantise(cb, &lsp_hz[i], wt, k, m, &se);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_lsps_scalar()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
From a vector of quantised LSP indexes, returns the quantised
(floating point) LSPs.
\*---------------------------------------------------------------------------*/
void decode_lsps_scalar(float lsp[], int indexes[], int order)
{
int i,k;
float lsp_hz[LPC_MAX];
const float * cb;
for(i=0; i<order; i++) {
k = lsp_cb[i].k;
cb = lsp_cb[i].cb;
lsp_hz[i] = cb[indexes[i]*k];
}
/* convert back to radians */
for(i=0; i<order; i++)
lsp[i] = (PI/4000.0)*lsp_hz[i];
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_lsps_diff_freq_vq()
AUTHOR......: David Rowe
DATE CREATED: 15 November 2011
Twenty-five bit LSP quantiser. LSPs 1-4 are quantised with scalar
LSP differences (in frequency, i.e difference from the previous
LSP). LSPs 5-10 are quantised with a VQ trained generated using
vqtrainjnd.c
\*---------------------------------------------------------------------------*/
void encode_lsps_diff_freq_vq(int indexes[], float lsp[], int order)
{
int i,k,m;
float lsp_hz[LPC_MAX];
float lsp__hz[LPC_MAX];
float dlsp[LPC_MAX];
float dlsp_[LPC_MAX];
float wt[LPC_MAX];
const float * cb;
float se;
for(i=0; i<LPC_ORD; i++) {
wt[i] = 1.0;
}
/* convert from radians to Hz so we can use human readable
frequencies */
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
/* scalar quantisers for LSP differences 1..4 */
wt[0] = 1.0;
for(i=0; i<4; i++) {
if (i)
dlsp[i] = lsp_hz[i] - lsp__hz[i-1];
else
dlsp[0] = lsp_hz[0];
k = lsp_cbd[i].k;
m = lsp_cbd[i].m;
cb = lsp_cbd[i].cb;
indexes[i] = quantise(cb, &dlsp[i], wt, k, m, &se);
dlsp_[i] = cb[indexes[i]*k];
if (i)
lsp__hz[i] = lsp__hz[i-1] + dlsp_[i];
else
lsp__hz[0] = dlsp_[0];
}
/* VQ LSPs 5,6,7,8,9,10 */
k = lsp_cbjnd[4].k;
m = lsp_cbjnd[4].m;
cb = lsp_cbjnd[4].cb;
indexes[4] = quantise(cb, &lsp_hz[4], &wt[4], k, m, &se);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_lsps_diff_freq_vq()
AUTHOR......: David Rowe
DATE CREATED: 15 Nov 2011
From a vector of quantised LSP indexes, returns the quantised
(floating point) LSPs.
\*---------------------------------------------------------------------------*/
void decode_lsps_diff_freq_vq(float lsp_[], int indexes[], int order)
{
int i,k,m;
float dlsp_[LPC_MAX];
float lsp__hz[LPC_MAX];
const float * cb;
/* scalar LSP differences */
for(i=0; i<4; i++) {
cb = lsp_cbd[i].cb;
dlsp_[i] = cb[indexes[i]];
if (i)
lsp__hz[i] = lsp__hz[i-1] + dlsp_[i];
else
lsp__hz[0] = dlsp_[0];
}
/* VQ */
k = lsp_cbjnd[4].k;
m = lsp_cbjnd[4].m;
cb = lsp_cbjnd[4].cb;
for(i=4; i<order; i++)
lsp__hz[i] = cb[indexes[4]*k+i-4];
/* convert back to radians */
for(i=0; i<order; i++)
lsp_[i] = (PI/4000.0)*lsp__hz[i];
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_lsps_diff_time()
AUTHOR......: David Rowe
DATE CREATED: 12 Sep 2012
Encode difference from preious frames's LSPs using
3,3,2,2,2,2,1,1,1,1 scalar quantisers (18 bits total).
\*---------------------------------------------------------------------------*/
void encode_lsps_diff_time(int indexes[],
float lsps[],
float lsps__prev[],
int order)
{
int i,k,m;
float lsps_dt[LPC_ORD];
float wt[LPC_MAX];
const float * cb;
float se;
/* Determine difference in time and convert from radians to Hz so
we can use human readable frequencies */
for(i=0; i<LPC_ORD; i++) {
lsps_dt[i] = (4000/PI)*(lsps[i] - lsps__prev[i]);
}
/* scalar quantisers */
wt[0] = 1.0;
for(i=0; i<order; i++) {
k = lsp_cbdt[i].k;
m = lsp_cbdt[i].m;
cb = lsp_cbdt[i].cb;
indexes[i] = quantise(cb, &lsps_dt[i], wt, k, m, &se);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_lsps_diff_time()
AUTHOR......: David Rowe
DATE CREATED: 15 Nov 2011
From a quantised LSP indexes, returns the quantised
(floating point) LSPs.
\*---------------------------------------------------------------------------*/
void decode_lsps_diff_time(
float lsps_[],
int indexes[],
float lsps__prev[],
int order)
{
int i,k,m;
const float * cb;
for(i=0; i<order; i++)
lsps_[i] = lsps__prev[i];
for(i=0; i<order; i++) {
k = lsp_cbdt[i].k;
cb = lsp_cbdt[i].cb;
lsps_[i] += (PI/4000.0)*cb[indexes[i]*k];
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_lsps_vq()
AUTHOR......: David Rowe
DATE CREATED: 15 Feb 2012
Multi-stage VQ LSP quantiser developed by Jean-Marc Valin.
\*---------------------------------------------------------------------------*/
void encode_lsps_vq(int *indexes, float *x, float *xq, int ndim)
{
int i, n1, n2, n3;
float err[LPC_ORD], err2[LPC_ORD], err3[LPC_ORD];
float w[LPC_ORD], w2[LPC_ORD], w3[LPC_ORD];
const float *codebook1 = lsp_cbjvm[0].cb;
const float *codebook2 = lsp_cbjvm[1].cb;
const float *codebook3 = lsp_cbjvm[2].cb;
assert(ndim <= LPC_ORD);
w[0] = MIN(x[0], x[1]-x[0]);
for (i=1;i<ndim-1;i++)
w[i] = MIN(x[i]-x[i-1], x[i+1]-x[i]);
w[ndim-1] = MIN(x[ndim-1]-x[ndim-2], PI-x[ndim-1]);
compute_weights(x, w, ndim);
n1 = find_nearest(codebook1, lsp_cbjvm[0].m, x, ndim);
for (i=0;i<ndim;i++)
{
xq[i] = codebook1[ndim*n1+i];
err[i] = x[i] - xq[i];
}
for (i=0;i<ndim/2;i++)
{
err2[i] = err[2*i];
err3[i] = err[2*i+1];
w2[i] = w[2*i];
w3[i] = w[2*i+1];
}
n2 = find_nearest_weighted(codebook2, lsp_cbjvm[1].m, err2, w2, ndim/2);
n3 = find_nearest_weighted(codebook3, lsp_cbjvm[2].m, err3, w3, ndim/2);
indexes[0] = n1;
indexes[1] = n2;
indexes[2] = n3;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_lsps_vq()
AUTHOR......: David Rowe
DATE CREATED: 15 Feb 2012
\*---------------------------------------------------------------------------*/
void decode_lsps_vq(int *indexes, float *xq, int ndim)
{
int i, n1, n2, n3;
const float *codebook1 = lsp_cbjvm[0].cb;
const float *codebook2 = lsp_cbjvm[1].cb;
const float *codebook3 = lsp_cbjvm[2].cb;
n1 = indexes[0];
n2 = indexes[1];
n3 = indexes[2];
for (i=0;i<ndim;i++)
{
xq[i] = codebook1[ndim*n1+i];
}
for (i=0;i<ndim/2;i++)
{
xq[2*i] += codebook2[ndim*n2/2+i];
xq[2*i+1] += codebook3[ndim*n3/2+i];
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: bw_expand_lsps()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Applies Bandwidth Expansion (BW) to a vector of LSPs. Prevents any
two LSPs getting too close together after quantisation. We know
from experiment that LSP quantisation errors < 12.5Hz (25Hz step
size) are inaudible so we use that as the minimum LSP separation.
\*---------------------------------------------------------------------------*/
void bw_expand_lsps(float lsp[],
int order
)
{
int i;
for(i=1; i<4; i++) {
if ((lsp[i] - lsp[i-1]) < 50*(PI/4000.0))
lsp[i] = lsp[i-1] + 50.0*(PI/4000.0);
}
/* As quantiser gaps increased, larger BW expansion was required
to prevent twinkly noises. This may need more experiment for
different quanstisers.
*/
for(i=4; i<order; i++) {
if (lsp[i] - lsp[i-1] < PI*(100.0/4000.0))
lsp[i] = lsp[i-1] + PI*(100.0/4000.0);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: locate_lsps_jnd_steps()
AUTHOR......: David Rowe
DATE CREATED: 27/10/2011
Applies a form of Bandwidth Expansion (BW) to a vector of LSPs.
Listening tests have determined that "quantising" the position of
each LSP to the non-linear steps below introduces a "just noticable
difference" in the synthesised speech.
This operation can be used before quantisation to limit the input
data to the quantiser to a number of discrete steps.
This operation can also be used during quantisation as a form of
hysteresis in the calculation of quantiser error. For example if
the quantiser target of lsp1 is 500 Hz, candidate vectors with lsp1
of 515 and 495 Hz sound effectively the same.
\*---------------------------------------------------------------------------*/
void locate_lsps_jnd_steps(float lsps[], int order)
{
int i;
float lsp_hz, step;
assert(order == 10);
/* quantise to 25Hz steps */
step = 25;
for(i=0; i<2; i++) {
lsp_hz = lsps[i]*4000.0/PI;
lsp_hz = floor(lsp_hz/step + 0.5)*step;
lsps[i] = lsp_hz*PI/4000.0;
if (i) {
if (lsps[i] == lsps[i-1])
lsps[i] += step*PI/4000.0;
}
}
/* quantise to 50Hz steps */
step = 50;
for(i=2; i<4; i++) {
lsp_hz = lsps[i]*4000.0/PI;
lsp_hz = floor(lsp_hz/step + 0.5)*step;
lsps[i] = lsp_hz*PI/4000.0;
if (i) {
if (lsps[i] == lsps[i-1])
lsps[i] += step*PI/4000.0;
}
}
/* quantise to 100Hz steps */
step = 100;
for(i=4; i<10; i++) {
lsp_hz = lsps[i]*4000.0/PI;
lsp_hz = floor(lsp_hz/step + 0.5)*step;
lsps[i] = lsp_hz*PI/4000.0;
if (i) {
if (lsps[i] == lsps[i-1])
lsps[i] += step*PI/4000.0;
}
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: apply_lpc_correction()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Apply first harmonic LPC correction at decoder. This helps improve
low pitch males after LPC modelling, like hts1a and morig.
\*---------------------------------------------------------------------------*/
void apply_lpc_correction(MODEL *model)
{
if (model->Wo < (PI*150.0/4000)) {
model->A[1] *= 0.032;
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_energy()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Encodes LPC energy using an E_LEVELS quantiser.
\*---------------------------------------------------------------------------*/
int encode_energy(float e)
{
int index;
float e_min = E_MIN_DB;
float e_max = E_MAX_DB;
float norm;
e = 10.0*log10(e);
norm = (e - e_min)/(e_max - e_min);
index = floor(E_LEVELS * norm + 0.5);
if (index < 0 ) index = 0;
if (index > (E_LEVELS-1)) index = E_LEVELS-1;
return index;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_energy()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Decodes energy using a E_LEVELS quantiser.
\*---------------------------------------------------------------------------*/
float decode_energy(int index)
{
float e_min = E_MIN_DB;
float e_max = E_MAX_DB;
float step;
float e;
step = (e_max - e_min)/E_LEVELS;
e = e_min + step*(index);
e = pow(10.0,e/10.0);
return e;
}
#ifdef NOT_USED
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_amplitudes()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Given the amplitude quantiser indexes recovers the harmonic
amplitudes.
\*---------------------------------------------------------------------------*/
float decode_amplitudes(kiss_fft_cfg fft_fwd_cfg,
MODEL *model,
float ak[],
int lsp_indexes[],
int energy_index,
float lsps[],
float *e
)
{
float snr;
decode_lsps_scalar(lsps, lsp_indexes, LPC_ORD);
bw_expand_lsps(lsps, LPC_ORD);
lsp_to_lpc(lsps, ak, LPC_ORD);
*e = decode_energy(energy_index);
aks_to_M2(ak, LPC_ORD, model, *e, &snr, 1, 0, 0, 1);
apply_lpc_correction(model);
return snr;
}
#endif
static float ge_coeff[2] = {0.8, 0.9};
void compute_weights2(const float *x, const float *xp, float *w, int ndim)
{
w[0] = 30;
w[1] = 1;
if (x[1]<0)
{
w[0] *= .6;
w[1] *= .3;
}
if (x[1]<-10)
{
w[0] *= .3;
w[1] *= .3;
}
/* Higher weight if pitch is stable */
if (fabs(x[0]-xp[0])<.2)
{
w[0] *= 2;
w[1] *= 1.5;
} else if (fabs(x[0]-xp[0])>.5) /* Lower if not stable */
{
w[0] *= .5;
}
/* Lower weight for low energy */
if (x[1] < xp[1]-10)
{
w[1] *= .5;
}
if (x[1] < xp[1]-20)
{
w[1] *= .5;
}
//w[0] = 30;
//w[1] = 1;
/* Square the weights because it's applied on the squared error */
w[0] *= w[0];
w[1] *= w[1];
}
/*---------------------------------------------------------------------------*\
FUNCTION....: quantise_WoE()
AUTHOR......: Jean-Marc Valin & David Rowe
DATE CREATED: 29 Feb 2012
Experimental joint Wo and LPC energy vector quantiser developed by
Jean-Marc Valin. Exploits correlations between the difference in
the log pitch and log energy from frame to frame. For example
both the pitch and energy tend to only change by small amounts
during voiced speech, however it is important that these changes be
coded carefully. During unvoiced speech they both change a lot but
the ear is less sensitve to errors so coarser quantisation is OK.
The ear is sensitive to log energy and loq pitch so we quantise in
these domains. That way the error measure used to quantise the
values is close to way the ear senses errors.
See http://jmspeex.livejournal.com/10446.html
\*---------------------------------------------------------------------------*/
void quantise_WoE(MODEL *model, float *e, float xq[])
{
int i, n1;
float x[2];
float err[2];
float w[2];
const float *codebook1 = ge_cb[0].cb;
int nb_entries = ge_cb[0].m;
int ndim = ge_cb[0].k;
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
x[0] = log10((model->Wo/PI)*4000.0/50.0)/log10(2);
x[1] = 10.0*log10(1e-4 + *e);
compute_weights2(x, xq, w, ndim);
for (i=0;i<ndim;i++)
err[i] = x[i]-ge_coeff[i]*xq[i];
n1 = find_nearest_weighted(codebook1, nb_entries, err, w, ndim);
for (i=0;i<ndim;i++)
{
xq[i] = ge_coeff[i]*xq[i] + codebook1[ndim*n1+i];
err[i] -= codebook1[ndim*n1+i];
}
/*
x = log2(4000*Wo/(PI*50));
2^x = 4000*Wo/(PI*50)
Wo = (2^x)*(PI*50)/4000;
*/
model->Wo = pow(2.0, xq[0])*(PI*50.0)/4000.0;
/* bit errors can make us go out of range leading to all sorts of
probs like seg faults */
if (model->Wo > Wo_max) model->Wo = Wo_max;
if (model->Wo < Wo_min) model->Wo = Wo_min;
model->L = PI/model->Wo; /* if we quantise Wo re-compute L */
*e = pow(10.0, xq[1]/10.0);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_WoE()
AUTHOR......: Jean-Marc Valin & David Rowe
DATE CREATED: 11 May 2012
Joint Wo and LPC energy vector quantiser developed my Jean-Marc
Valin. Returns index, and updated states xq[].
\*---------------------------------------------------------------------------*/
int encode_WoE(MODEL *model, float e, float xq[])
{
int i, n1;
float x[2];
float err[2];
float w[2];
const float *codebook1 = ge_cb[0].cb;
int nb_entries = ge_cb[0].m;
int ndim = ge_cb[0].k;
assert((1<<WO_E_BITS) == nb_entries);
if (e < 0.0) e = 0; /* occasional small negative energies due LPC round off I guess */
x[0] = log10((model->Wo/PI)*4000.0/50.0)/log10(2);
x[1] = 10.0*log10(1e-4 + e);
compute_weights2(x, xq, w, ndim);
for (i=0;i<ndim;i++)
err[i] = x[i]-ge_coeff[i]*xq[i];
n1 = find_nearest_weighted(codebook1, nb_entries, err, w, ndim);
for (i=0;i<ndim;i++)
{
xq[i] = ge_coeff[i]*xq[i] + codebook1[ndim*n1+i];
err[i] -= codebook1[ndim*n1+i];
}
//printf("enc: %f %f (%f)(%f) \n", xq[0], xq[1], e, 10.0*log10(1e-4 + e));
return n1;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_WoE()
AUTHOR......: Jean-Marc Valin & David Rowe
DATE CREATED: 11 May 2012
Joint Wo and LPC energy vector quantiser developed my Jean-Marc
Valin. Given index and states xq[], returns Wo & E, and updates
states xq[].
\*---------------------------------------------------------------------------*/
void decode_WoE(MODEL *model, float *e, float xq[], int n1)
{
int i;
float err[2];
const float *codebook1 = ge_cb[0].cb;
int ndim = ge_cb[0].k;
float Wo_min = TWO_PI/P_MAX;
float Wo_max = TWO_PI/P_MIN;
for (i=0;i<ndim;i++)
{
xq[i] = ge_coeff[i]*xq[i] + codebook1[ndim*n1+i];
err[i] -= codebook1[ndim*n1+i];
}
//printf("dec: %f %f\n", xq[0], xq[1]);
model->Wo = pow(2.0, xq[0])*(PI*50.0)/4000.0;
/* bit errors can make us go out of range leading to all sorts of
probs like seg faults */
if (model->Wo > Wo_max) model->Wo = Wo_max;
if (model->Wo < Wo_min) model->Wo = Wo_min;
model->L = PI/model->Wo; /* if we quantise Wo re-compute L */
*e = pow(10.0, xq[1]/10.0);
}
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