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Working codec2 support

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This commit is contained in:
Brian West
2010-09-21 12:35:41 -05:00
parent 549b2356d6
commit 62b55523c2
150 changed files with 14078 additions and 3 deletions
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53
libs/libcodec2/src/Makefile.am Normal file
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80
libs/libcodec2/src/c2dec.c Normal file
View File

@@ -0,0 +1,80 @@
/*---------------------------------------------------------------------------*\
FILE........: c2dec.c
AUTHOR......: David Rowe
DATE CREATED: 23/8/2010
Decodes a file of bits to a file of raw speech samples using codec2. Demo
program for codec2.
NOTE: the bit file is not packed, 51 bits/frame actually consumes 51
bytes/frame on disk. If you are using this for a real world
application you may want to pack the 51 bytes into 7 bytes.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2010 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "codec2.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
int main(int argc, char *argv[])
{
static const int bitsSize = ((CODEC2_BITS_PER_FRAME + 7) / 8);
void *codec2;
FILE *fin;
FILE *fout;
short buf[CODEC2_SAMPLES_PER_FRAME];
unsigned char bits[bitsSize];
if (argc != 3) {
printf("usage: %s InputBitFile OutputRawSpeechFile\n", argv[0]);
exit(1);
}
if ( (fin = fopen(argv[1],"rb")) == NULL ) {
fprintf(stderr, "Error opening input bit file: %s: %s.\n",
argv[1], strerror(errno));
exit(1);
}
if ( (fout = fopen(argv[2],"wb")) == NULL ) {
fprintf(stderr, "Error opening output speech file: %s: %s.\n",
argv[2], strerror(errno));
exit(1);
}
codec2 = codec2_create();
while(fread(bits, sizeof(char), bitsSize, fin) == bitsSize) {
codec2_decode(codec2, buf, bits);
fwrite(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fout);
}
codec2_destroy(codec2);
fclose(fin);
fclose(fout);
return 0;
}

82
libs/libcodec2/src/c2enc.c Normal file
View File

@@ -0,0 +1,82 @@
/*---------------------------------------------------------------------------*\
FILE........: c2enc.c
AUTHOR......: David Rowe
DATE CREATED: 23/8/2010
Encodes a file of raw speech samples using codec2 and ouputs a file
of bits (each bit is stored in the LSB or each output byte). Demo
program for codec2.
NOTE: the bit file is not packed, 51 bits/frame actually consumes 51
bytes/frame on disk. If you are using this for a real world
application you may want to pack the 51 bytes into 7 bytes.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2010 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "codec2.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
int main(int argc, char *argv[])
{
static const int bitsSize = ((CODEC2_BITS_PER_FRAME + 7) / 8);
void *codec2;
FILE *fin;
FILE *fout;
short buf[CODEC2_SAMPLES_PER_FRAME];
unsigned char bits[bitsSize];
if (argc != 3) {
printf("usage: %s InputRawspeechFile OutputBitFile\n", argv[0]);
exit(1);
}
if ( (fin = fopen(argv[1],"rb")) == NULL ) {
fprintf(stderr, "Error opening input bit file: %s: %s.\n",
argv[1], strerror(errno));
exit(1);
}
if ( (fout = fopen(argv[2],"wb")) == NULL ) {
fprintf(stderr, "Error opening output speech file: %s: %s.\n",
argv[2], strerror(errno));
exit(1);
}
codec2 = codec2_create();
while(fread(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fin) ==
CODEC2_SAMPLES_PER_FRAME) {
codec2_encode(codec2, bits, buf);
fwrite(bits, sizeof(char), bitsSize, fout);
}
codec2_destroy(codec2);
fclose(fin);
fclose(fout);
return 0;
}

408
libs/libcodec2/src/c2sim.c Normal file
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@@ -0,0 +1,408 @@
/*---------------------------------------------------------------------------*\
FILE........: c2sim.c
AUTHOR......: David Rowe
DATE CREATED: 20/8/2010
Codec2 simulation. Combines encoder and decoder and allows switching in
out various algorithms and quantisation steps.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <math.h>
#include "defines.h"
#include "sine.h"
#include "nlp.h"
#include "dump.h"
#include "lpc.h"
#include "lsp.h"
#include "quantise.h"
#include "phase.h"
#include "postfilter.h"
#include "interp.h"
/*---------------------------------------------------------------------------*\
switch_present()
Searches the command line arguments for a "switch". If the switch is
found, returns the command line argument where it ws found, else returns
NULL.
\*---------------------------------------------------------------------------*/
int switch_present(sw,argc,argv)
register char sw[]; /* switch in string form */
register int argc; /* number of command line arguments */
register char *argv[]; /* array of command line arguments in string form */
{
register int i; /* loop variable */
for(i=1; i<argc; i++)
if (!strcmp(sw,argv[i]))
return(i);
return 0;
}
void synth_one_frame(short buf[], MODEL *model, float Sn_[], float Pn[]);
/*---------------------------------------------------------------------------*\
MAIN
\*---------------------------------------------------------------------------*/
int main(int argc, char *argv[])
{
FILE *fout; /* output speech file */
FILE *fin; /* input speech file */
short buf[N]; /* input/output buffer */
float Sn[M]; /* float input speech samples */
COMP Sw[FFT_ENC]; /* DFT of Sn[] */
float w[M]; /* time domain hamming window */
COMP W[FFT_ENC]; /* DFT of w[] */
MODEL model;
float Pn[2*N]; /* trapezoidal synthesis window */
float Sn_[2*N]; /* synthesised speech */
int i; /* loop variable */
int frames;
float prev_Wo;
float pitch;
int voiced1;
char out_file[MAX_STR];
int arg;
float snr;
float sum_snr;
int lpc_model, order;
int lsp, lsp_quantiser;
float ak[LPC_MAX];
COMP Sw_[FFT_ENC];
int dump;
int phase0;
float ex_phase[MAX_AMP+1];
int postfilt;
float bg_est;
int hand_voicing;
FILE *fvoicing;
MODEL prev_model, interp_model;
int decimate;
void *nlp_states;
for(i=0; i<M; i++)
Sn[i] = 1.0;
for(i=0; i<2*N; i++)
Sn_[i] = 0;
prev_Wo = TWO_PI/P_MAX;
prev_model.Wo = TWO_PI/P_MIN;
prev_model.L = floor(PI/prev_model.Wo);
for(i=1; i<=prev_model.L; i++) {
prev_model.A[i] = 0.0;
prev_model.phi[i] = 0.0;
}
for(i=1; i<=MAX_AMP; i++) {
ex_phase[i] = 0.0;
}
nlp_states = nlp_create();
if (argc < 2) {
fprintf(stderr, "\nCodec2 - 2400 bit/s speech codec - Simulation Program\n"
"\thttp://rowetel.com/codec2.html\n\n"
"usage: %s InputFile [-o OutputFile]\n"
"\t[-o lpc Order]\n"
"\t[--lsp]\n"
"\t[--phase0]\n"
"\t[--postfilter]\n"
"\t[--hand_voicing]\n"
"\t[--dec]\n"
"\t[--dump DumpFilePrefix]\n", argv[0]);
exit(1);
}
/* Interpret command line arguments -------------------------------------*/
/* Input file */
if ((fin = fopen(argv[1],"rb")) == NULL) {
fprintf(stderr, "Error opening input bit file: %s: %s.\n",
argv[1], strerror(errno));
exit(1);
}
/* Output file */
if ((arg = switch_present("-o",argc,argv))) {
if ((fout = fopen(argv[arg+1],"wb")) == NULL) {
fprintf(stderr, "Error opening output speech file: %s: %s.\n",
argv[arg+1], strerror(errno));
exit(1);
}
strcpy(out_file,argv[arg+1]);
}
else
fout = NULL;
lpc_model = 0;
if ((arg = switch_present("--lpc",argc,argv))) {
lpc_model = 1;
order = atoi(argv[arg+1]);
if ((order < 4) || (order > 20)) {
fprintf(stderr, "Error in lpc order: %d\n", order);
exit(1);
}
}
dump = switch_present("--dump",argc,argv);
if (dump)
dump_on(argv[dump+1]);
lsp = switch_present("--lsp",argc,argv);
lsp_quantiser = 0;
phase0 = switch_present("--phase0",argc,argv);
if (phase0) {
ex_phase[0] = 0;
}
hand_voicing = switch_present("--hand_voicing",argc,argv);
if (hand_voicing) {
fvoicing = fopen(argv[hand_voicing+1],"rt");
assert(fvoicing != NULL);
}
bg_est = 0.0;
postfilt = switch_present("--postfilter",argc,argv);
decimate = switch_present("--dec",argc,argv);
/* Initialise ------------------------------------------------------------*/
make_analysis_window(w,W);
make_synthesis_window(Pn);
quantise_init();
/* Main loop ------------------------------------------------------------*/
frames = 0;
sum_snr = 0;
while(fread(buf,sizeof(short),N,fin)) {
frames++;
/* Read input speech */
for(i=0; i<M-N; i++)
Sn[i] = Sn[i+N];
for(i=0; i<N; i++)
Sn[i+M-N] = buf[i];
/* Estimate pitch */
nlp(nlp_states,Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&prev_Wo);
prev_Wo = TWO_PI/pitch;
model.Wo = TWO_PI/pitch;
/* estimate model parameters */
dft_speech(Sw, Sn, w);
two_stage_pitch_refinement(&model, Sw);
estimate_amplitudes(&model, Sw, W);
dump_Sn(Sn); dump_Sw(Sw); dump_model(&model);
/* optional zero-phase modelling */
if (phase0) {
float Wn[M]; /* windowed speech samples */
float Rk[LPC_ORD+1]; /* autocorrelation coeffs */
dump_phase(&model.phi[0], model.L);
/* find aks here, these are overwritten if LPC modelling is enabled */
for(i=0; i<M; i++)
Wn[i] = Sn[i]*w[i];
autocorrelate(Wn,Rk,M,LPC_ORD);
levinson_durbin(Rk,ak,LPC_ORD);
if (lpc_model)
assert(order == LPC_ORD);
dump_ak(ak, LPC_ORD);
/* determine voicing */
snr = est_voicing_mbe(&model, Sw, W, (FS/TWO_PI)*model.Wo, Sw_);
dump_Sw_(Sw_);
dump_snr(snr);
/* just to make sure we are not cheating - kill all phases */
for(i=0; i<MAX_AMP; i++)
model.phi[i] = 0;
if (hand_voicing) {
fscanf(fvoicing,"%d\n",&model.voiced);
}
}
/* optional LPC model amplitudes */
if (lpc_model) {
int lpc_correction;
float e;
float lsps[LPC_ORD];
int lsp_indexes[LPC_ORD];
e = speech_to_uq_lsps(lsps, ak, Sn, w, order);
lpc_correction = need_lpc_correction(&model, ak, e);
if (lsp) {
encode_lsps(lsp_indexes, lsps, LPC_ORD);
/*
for(i=0; i<LPC_ORD; i++)
printf("lsps[%d] = %f lsp_indexes[%d] = %d\n",
i, lsps[i], i, lsp_indexes[i]);
printf("\n");
*/
decode_lsps(lsps, lsp_indexes, LPC_ORD);
bw_expand_lsps(lsps, LPC_ORD);
lsp_to_lpc(lsps, ak, LPC_ORD);
}
e = decode_energy(encode_energy(e));
model.Wo = decode_Wo(encode_Wo(model.Wo));
aks_to_M2(ak, order, &model, e, &snr, 1);
apply_lpc_correction(&model, lpc_correction);
sum_snr += snr;
dump_quantised_model(&model);
}
/* option decimation to 20ms rate, which enables interpolation
routine to synthesise in between frame */
if (decimate) {
if (!phase0) {
printf("needs --phase0 to resample phase for interpolated Wo\n");
exit(0);
}
/* odd frame - interpolate */
if (frames%2) {
#ifdef TEST
model.voiced = 1;
prev_model.voiced = 1;
if (fabs(prev_model.Wo - model.Wo) < 0.1*model.Wo) {
interp_model.voiced = 1;
interpolate(&interp_model, &prev_model, &model);
for(i=0; i<=interp_model.L; i++) {
interp_model.phi[i] = phi1[i];
}
printf("interp\n");
}
else
interp_model = tmp_model;
#endif
interp_model.voiced = voiced1;
interpolate(&interp_model, &prev_model, &model);
if (phase0)
phase_synth_zero_order(&interp_model, ak, ex_phase);
if (postfilt)
postfilter(&interp_model, &bg_est);
synth_one_frame(buf, &interp_model, Sn_, Pn);
if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
if (phase0)
phase_synth_zero_order(&model, ak, ex_phase);
if (postfilt)
postfilter(&model, &bg_est);
synth_one_frame(buf, &model, Sn_, Pn);
if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
prev_model = model;
}
else {
voiced1 = model.voiced;
}
}
else {
if (phase0)
phase_synth_zero_order(&model, ak, ex_phase);
if (postfilt)
postfilter(&model, &bg_est);
synth_one_frame(buf, &model, Sn_, Pn);
if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
}
}
if (fout != NULL)
fclose(fout);
if (lpc_model)
printf("SNR av = %5.2f dB\n", sum_snr/frames);
if (dump)
dump_off();
if (hand_voicing)
fclose(fvoicing);
nlp_destroy(nlp_states);
return 0;
}
void synth_one_frame(short buf[], MODEL *model, float Sn_[], float Pn[])
{
int i;
synthesise(Sn_, model, Pn, 1);
for(i=0; i<N; i++) {
if (Sn_[i] > 32767.0)
buf[i] = 32767;
else if (Sn_[i] < -32767.0)
buf[i] = -32767;
else
buf[i] = Sn_[i];
}
}

9
libs/libcodec2/src/codeall.sh Executable file
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#!/bin/sh
# codeall.sh
# David Rowe 24 sep 2009
# Code all samples using various processing steps
./code.sh hts1a
./code.sh hts2a
./code.sh mmt1
./code.sh morig
./code.sh forig

162
libs/libcodec2/src/codebook.c Normal file
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float codebook_lsp1[] = {
225,
250,
275,
300,
325,
350,
375,
400,
425,
450,
475,
500,
525,
550,
575,
600,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp2[] = {
325,
350,
375,
400,
425,
450,
475,
500,
525,
550,
575,
600,
625,
650,
675,
700,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp3[] = {
500,
550,
600,
650,
700,
750,
800,
850,
900,
950,
1000,
1050,
1100,
1150,
1200,
1250,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp4[] = {
700,
800,
900,
1000,
1100,
1200,
1300,
1400,
1500,
1600,
1700,
1800,
1900,
2000,
2100,
2200,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp5[] = {
950,
1050,
1150,
1250,
1350,
1450,
1550,
1650,
1750,
1850,
1950,
2050,
2150,
2250,
2350,
2450,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp6[] = {
1100,
1200,
1300,
1400,
1500,
1600,
1700,
1800,
1900,
2000,
2100,
2200,
2300,
2400,
2500,
2600,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp7[] = {
1500,
1600,
1700,
1800,
1900,
2000,
2100,
2200,
2300,
2400,
2500,
2600,
2700,
2800,
2900,
3000,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp8[] = {
2300,
2400,
2500,
2600,
2700,
2800,
2900,
3000,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp9[] = {
2500,
2600,
2700,
2800,
2900,
3000,
3100,
3200,0,0,0,0,0,0,0,0,0,0
};
float codebook_lsp10[] = {
2900,
3100,
3300,
3500,0,0,0,0,0,0,0,0,0,0
};

15
libs/libcodec2/src/codebook.h Normal file
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#ifndef CODEBOOK_H
#define CODEBOOK_H
extern float codebook_lsp1[];
extern float codebook_lsp2[];
extern float codebook_lsp3[];
extern float codebook_lsp4[];
extern float codebook_lsp5[];
extern float codebook_lsp6[];
extern float codebook_lsp7[];
extern float codebook_lsp8[];
extern float codebook_lsp9[];
extern float codebook_lsp10[];
#endif

337
libs/libcodec2/src/codec2.c Normal file
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/*---------------------------------------------------------------------------*\
FILE........: codec2.c
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Codec2 fully quantised encoder and decoder functions. If you want use
codec2, the codec2_xxx functions are for you.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2010 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "defines.h"
#include "sine.h"
#include "nlp.h"
#include "dump.h"
#include "lpc.h"
#include "quantise.h"
#include "phase.h"
#include "interp.h"
#include "postfilter.h"
#include "codec2.h"
typedef struct {
float Sn[M]; /* input speech */
float w[M]; /* time domain hamming window */
COMP W[FFT_ENC]; /* DFT of w[] */
float Pn[2*N]; /* trapezoidal synthesis window */
float Sn_[2*N]; /* synthesised speech */
float prev_Wo; /* previous frame's pitch estimate */
float ex_phase; /* excitation model phase track */
float bg_est; /* background noise estimate for post filter */
MODEL prev_model; /* model parameters from 20ms ago */
void *nlp; /* pitch predictor states */
} CODEC2;
/*---------------------------------------------------------------------------*\
FUNCTION HEADERS
\*---------------------------------------------------------------------------*/
void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[]);
void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model,float ak[]);
/*---------------------------------------------------------------------------*\
FUNCTIONS
\*---------------------------------------------------------------------------*/
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_create
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Create and initialise an instance of the codec. Returns a pointer
to the codec states or NULL on failure. One set of states is
sufficient for a full duuplex codec (i.e. an encoder and decoder).
You don't need separate states for encoders and decoders. See
c2enc.c and c2dec.c for examples.
\*---------------------------------------------------------------------------*/
void *codec2_create()
{
CODEC2 *c2;
int i,l;
c2 = (CODEC2*)malloc(sizeof(CODEC2));
if (c2 == NULL)
return NULL;
for(i=0; i<M; i++)
c2->Sn[i] = 1.0;
for(i=0; i<2*N; i++)
c2->Sn_[i] = 0;
make_analysis_window(c2->w,c2->W);
make_synthesis_window(c2->Pn);
quantise_init();
c2->prev_Wo = 0.0;
c2->bg_est = 0.0;
c2->ex_phase = 0.0;
for(l=1; l<=MAX_AMP; l++)
c2->prev_model.A[l] = 0.0;
c2->prev_model.Wo = TWO_PI/P_MAX;
c2->nlp = nlp_create();
if (c2->nlp == NULL) {
free (c2);
return NULL;
}
return (void*)c2;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_create
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Destroy an instance of the codec.
\*---------------------------------------------------------------------------*/
void codec2_destroy(void *codec2_state)
{
CODEC2 *c2;
assert(codec2_state != NULL);
c2 = (CODEC2*)codec2_state;
nlp_destroy(c2->nlp);
free(codec2_state);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_encode
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Encodes 160 speech samples (20ms of speech) into 51 bits.
The codec2 algorithm actually operates internally on 10ms (80
sample) frames, so we run the encoding algorithm twice. On the
first frame we just send the voicing bit. One the second frame we
send all model parameters.
The bit allocation is:
Parameter bits/frame
--------------------------------------
Harmonic magnitudes (LSPs) 36
Low frequency LPC correction 1
Energy 5
Wo (fundamental frequnecy) 7
Voicing (10ms update) 2
TOTAL 51
\*---------------------------------------------------------------------------*/
void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
{
CODEC2 *c2;
MODEL model;
int voiced1, voiced2;
int lsp_indexes[LPC_ORD];
int lpc_correction;
int energy_index;
int Wo_index;
int i;
unsigned int nbit = 0;
assert(codec2_state != NULL);
c2 = (CODEC2*)codec2_state;
/* first 10ms analysis frame - we just want voicing */
analyse_one_frame(c2, &model, speech);
voiced1 = model.voiced;
/* second 10ms analysis frame */
analyse_one_frame(c2, &model, &speech[N]);
voiced2 = model.voiced;
Wo_index = encode_Wo(model.Wo);
encode_amplitudes(lsp_indexes,
&lpc_correction,
&energy_index,
&model,
c2->Sn,
c2->w);
memset(bits, '\0', ((CODEC2_BITS_PER_FRAME + 7) / 8));
pack(bits, &nbit, Wo_index, WO_BITS);
for(i=0; i<LPC_ORD; i++) {
pack(bits, &nbit, lsp_indexes[i], lsp_bits(i));
}
pack(bits, &nbit, lpc_correction, 1);
pack(bits, &nbit, energy_index, E_BITS);
pack(bits, &nbit, voiced1, 1);
pack(bits, &nbit, voiced2, 1);
assert(nbit == CODEC2_BITS_PER_FRAME);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_decode
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Decodes frames of 51 bits into 160 samples (20ms) of speech.
\*---------------------------------------------------------------------------*/
void codec2_decode(void *codec2_state, short speech[],
const unsigned char * bits)
{
CODEC2 *c2;
MODEL model;
int voiced1, voiced2;
int lsp_indexes[LPC_ORD];
int lpc_correction;
int energy_index;
int Wo_index;
float ak[LPC_ORD+1];
int i;
unsigned int nbit = 0;
MODEL model_interp;
assert(codec2_state != NULL);
c2 = (CODEC2*)codec2_state;
Wo_index = unpack(bits, &nbit, WO_BITS);
for(i=0; i<LPC_ORD; i++) {
lsp_indexes[i] = unpack(bits, &nbit, lsp_bits(i));
}
lpc_correction = unpack(bits, &nbit, 1);
energy_index = unpack(bits, &nbit, E_BITS);
voiced1 = unpack(bits, &nbit, 1);
voiced2 = unpack(bits, &nbit, 1);
assert(nbit == CODEC2_BITS_PER_FRAME);
model.Wo = decode_Wo(Wo_index);
model.L = PI/model.Wo;
decode_amplitudes(&model,
ak,
lsp_indexes,
lpc_correction,
energy_index);
model.voiced = voiced2;
model_interp.voiced = voiced1;
interpolate(&model_interp, &c2->prev_model, &model);
synthesise_one_frame(c2, speech, &model_interp, ak);
synthesise_one_frame(c2, &speech[N], &model, ak);
memcpy(&c2->prev_model, &model, sizeof(MODEL));
}
/*---------------------------------------------------------------------------*\
FUNCTION....: synthesise_one_frame()
AUTHOR......: David Rowe
DATE CREATED: 23/8/2010
Synthesise 80 speech samples (10ms) from model parameters.
\*---------------------------------------------------------------------------*/
void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model, float ak[])
{
int i;
phase_synth_zero_order(model, ak, &c2->ex_phase);
postfilter(model, &c2->bg_est);
synthesise(c2->Sn_, model, c2->Pn, 1);
for(i=0; i<N; i++) {
if (c2->Sn_[i] > 32767.0)
speech[i] = 32767;
else if (c2->Sn_[i] < -32767.0)
speech[i] = -32767;
else
speech[i] = c2->Sn_[i];
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: analyse_one_frame()
AUTHOR......: David Rowe
DATE CREATED: 23/8/2010
Extract sinusoidal model parameters from 80 speech samples (10ms of
speech).
\*---------------------------------------------------------------------------*/
void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[])
{
COMP Sw[FFT_ENC];
COMP Sw_[FFT_ENC];
float pitch;
int i;
/* Read input speech */
for(i=0; i<M-N; i++)
c2->Sn[i] = c2->Sn[i+N];
for(i=0; i<N; i++)
c2->Sn[i+M-N] = speech[i];
dft_speech(Sw, c2->Sn, c2->w);
/* Estimate pitch */
nlp(c2->nlp,c2->Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&c2->prev_Wo);
c2->prev_Wo = TWO_PI/pitch;
model->Wo = TWO_PI/pitch;
model->L = PI/model->Wo;
/* estimate model parameters */
dft_speech(Sw, c2->Sn, c2->w);
two_stage_pitch_refinement(model, Sw);
estimate_amplitudes(model, Sw, c2->W);
est_voicing_mbe(model, Sw, c2->W, (FS/TWO_PI)*model->Wo, Sw_);
}

43
libs/libcodec2/src/codec2.h Normal file
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@@ -0,0 +1,43 @@
/*---------------------------------------------------------------------------*\
FILE........: codec2.h
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
Codec2 fully quantised encoder and decoder functions. If you want use
codec2, these are the functions you need to call.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2010 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __CODEC2__
#define __CODEC2__
#include "codebook.h"
#define CODEC2_SAMPLES_PER_FRAME 160
#define CODEC2_BITS_PER_FRAME 51
void *codec2_create();
void codec2_destroy(void *codec2_state);
void codec2_encode(void *codec2_state, unsigned char * bits, short speech_in[]);
void codec2_decode(void *codec2_state, short speech_out[],
const unsigned char * bits);
#endif

39
libs/libcodec2/src/comp.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: comp.h
AUTHOR......: David Rowe
DATE CREATED: 24/08/09
Complex number definition.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __COMP__
#define __COMP__
/* Complex number */
typedef struct {
float real;
float imag;
} COMP;
#endif

84
libs/libcodec2/src/defines.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: defines.h
AUTHOR......: David Rowe
DATE CREATED: 23/4/93
Defines and structures used throughout the codec.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __DEFINES__
#define __DEFINES__
/*---------------------------------------------------------------------------*\
DEFINES
\*---------------------------------------------------------------------------*/
/* General defines */
#define N 80 /* number of samples per frame */
#define MAX_AMP 80 /* maximum number of harmonics */
#define PI 3.141592654 /* mathematical constant */
#define TWO_PI 6.283185307 /* mathematical constant */
#define FS 8000 /* sample rate in Hz */
#define MAX_STR 256 /* maximum string size */
#define NW 279 /* analysis window size */
#define FFT_ENC 512 /* size of FFT used for encoder */
#define FFT_DEC 512 /* size of FFT used in decoder */
#define TW 40 /* Trapezoidal synthesis window overlap */
#define V_THRESH 4.0 /* voicing threshold in dB */
#define LPC_MAX 20 /* maximum LPC order */
#define LPC_ORD 10 /* phase modelling LPC order */
/* Pitch estimation defines */
#define M 320 /* pitch analysis frame size */
#define P_MIN 20 /* minimum pitch */
#define P_MAX 160 /* maximum pitch */
/*---------------------------------------------------------------------------*\
TYPEDEFS
\*---------------------------------------------------------------------------*/
/* Complex number */
typedef struct {
float real;
float imag;
} COMP;
/* Structure to hold model parameters for one frame */
typedef struct {
float Wo; /* fundamental frequency estimate in radians */
int L; /* number of harmonics */
float A[MAX_AMP]; /* amplitiude of each harmonic */
float phi[MAX_AMP]; /* phase of each harmonic */
int voiced; /* non-zero if this frame is voiced */
} MODEL;
#endif

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libs/libcodec2/src/dump.c Normal file
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/*---------------------------------------------------------------------------*\
FILE........: dump.c
AUTHOR......: David Rowe
DATE CREATED: 25/8/09
Routines to dump data to text files for Octave analysis.
\*---------------------------------------------------------------------------*/
/*
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "defines.h"
#include "dump.h"
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
static int dumpon = 0;
static FILE *fsn = NULL;
static FILE *fsw = NULL;
static FILE *fsw_ = NULL;
static FILE *fmodel = NULL;
static FILE *fqmodel = NULL;
static FILE *fpw = NULL;
static FILE *flsp = NULL;
static FILE *fphase = NULL;
static FILE *fphase_ = NULL;
static FILE *ffw = NULL;
static FILE *fe = NULL;
static FILE *fsq = NULL;
static FILE *fdec = NULL;
static FILE *fsnr = NULL;
static FILE *fak = NULL;
static FILE *fbg = NULL;
static FILE *fE = NULL;
static char prefix[MAX_STR];
void dump_on(char p[]) {
dumpon = 1;
strcpy(prefix, p);
}
void dump_off(){
if (fsn != NULL)
fclose(fsn);
if (fsw != NULL)
fclose(fsw);
if (fsw_ != NULL)
fclose(fsw_);
if (fmodel != NULL)
fclose(fmodel);
if (fqmodel != NULL)
fclose(fqmodel);
if (fpw != NULL)
fclose(fpw);
if (flsp != NULL)
fclose(flsp);
if (fphase != NULL)
fclose(fphase);
if (fphase_ != NULL)
fclose(fphase_);
if (ffw != NULL)
fclose(ffw);
if (fe != NULL)
fclose(fe);
if (fsq != NULL)
fclose(fsq);
if (fdec != NULL)
fclose(fdec);
if (fsnr != NULL)
fclose(fsnr);
if (fak != NULL)
fclose(fak);
if (fbg != NULL)
fclose(fbg);
if (fE != NULL)
fclose(fE);
}
void dump_Sn(float Sn[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fsn == NULL) {
sprintf(s,"%s_sn.txt", prefix);
fsn = fopen(s, "wt");
assert(fsn != NULL);
}
/* split across two lines to avoid max line length problems */
/* reconstruct in Octave */
for(i=0; i<M/2; i++)
fprintf(fsn,"%f\t",Sn[i]);
fprintf(fsn,"\n");
for(i=M/2; i<M; i++)
fprintf(fsn,"%f\t",Sn[i]);
fprintf(fsn,"\n");
}
void dump_Sw(COMP Sw[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fsw == NULL) {
sprintf(s,"%s_sw.txt", prefix);
fsw = fopen(s, "wt");
assert(fsw != NULL);
}
for(i=0; i<FFT_ENC/2; i++)
fprintf(fsw,"%f\t",
10.0*log10(Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag));
fprintf(fsw,"\n");
}
void dump_Sw_(COMP Sw_[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fsw_ == NULL) {
sprintf(s,"%s_sw_.txt", prefix);
fsw_ = fopen(s, "wt");
assert(fsw_ != NULL);
}
for(i=0; i<FFT_ENC/2; i++)
fprintf(fsw_,"%f\t",
10.0*log10(Sw_[i].real*Sw_[i].real + Sw_[i].imag*Sw_[i].imag));
fprintf(fsw_,"\n");
}
void dump_model(MODEL *model) {
int l;
char s[MAX_STR];
if (!dumpon) return;
if (fmodel == NULL) {
sprintf(s,"%s_model.txt", prefix);
fmodel = fopen(s, "wt");
assert(fmodel != NULL);
}
fprintf(fmodel,"%f\t%d\t", model->Wo, model->L);
for(l=1; l<=model->L; l++)
fprintf(fmodel,"%f\t",model->A[l]);
for(l=model->L+1; l<MAX_AMP; l++)
fprintf(fmodel,"0.0\t");
fprintf(fmodel,"%d\t",model->voiced);
fprintf(fmodel,"\n");
}
void dump_quantised_model(MODEL *model) {
int l;
char s[MAX_STR];
if (!dumpon) return;
if (fqmodel == NULL) {
sprintf(s,"%s_qmodel.txt", prefix);
fqmodel = fopen(s, "wt");
assert(fqmodel != NULL);
}
fprintf(fqmodel,"%f\t%d\t", model->Wo, model->L);
for(l=1; l<=model->L; l++)
fprintf(fqmodel,"%f\t",model->A[l]);
for(l=model->L+1; l<MAX_AMP; l++)
fprintf(fqmodel,"0.0\t");
fprintf(fqmodel,"\n");
}
void dump_phase(float phase[], int L) {
int l;
char s[MAX_STR];
if (!dumpon) return;
if (fphase == NULL) {
sprintf(s,"%s_phase.txt", prefix);
fphase = fopen(s, "wt");
assert(fphase != NULL);
}
for(l=1; l<=L; l++)
fprintf(fphase,"%f\t",phase[l]);
for(l=L+1; l<MAX_AMP; l++)
fprintf(fphase,"%f\t",0.0);
fprintf(fphase,"\n");
}
void dump_phase_(float phase_[], int L) {
int l;
char s[MAX_STR];
if (!dumpon) return;
if (fphase_ == NULL) {
sprintf(s,"%s_phase_.txt", prefix);
fphase_ = fopen(s, "wt");
assert(fphase_ != NULL);
}
for(l=1; l<=L; l++)
fprintf(fphase_,"%f\t",phase_[l]);
for(l=L+1; l<MAX_AMP; l++)
fprintf(fphase_,"%f\t",0.0);
fprintf(fphase_,"\n");
}
void dump_snr(float snr) {
char s[MAX_STR];
if (!dumpon) return;
if (fsnr == NULL) {
sprintf(s,"%s_snr.txt", prefix);
fsnr = fopen(s, "wt");
assert(fsnr != NULL);
}
fprintf(fsnr,"%f\n",snr);
}
void dump_Pw(COMP Pw[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fpw == NULL) {
sprintf(s,"%s_pw.txt", prefix);
fpw = fopen(s, "wt");
assert(fpw != NULL);
}
for(i=0; i<FFT_DEC/2; i++)
fprintf(fpw,"%f\t",Pw[i].real);
fprintf(fpw,"\n");
}
void dump_lsp(float lsp[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (flsp == NULL) {
sprintf(s,"%s_lsp.txt", prefix);
flsp = fopen(s, "wt");
assert(flsp != NULL);
}
for(i=0; i<10; i++)
fprintf(flsp,"%f\t",lsp[i]);
fprintf(flsp,"\n");
}
void dump_ak(float ak[], int order) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fak == NULL) {
sprintf(s,"%s_ak.txt", prefix);
fak = fopen(s, "wt");
assert(fak != NULL);
}
for(i=0; i<=order; i++)
fprintf(fak,"%f\t",ak[i]);
fprintf(fak,"\n");
}
void dump_Fw(COMP Fw[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (ffw == NULL) {
sprintf(s,"%s_fw.txt", prefix);
ffw = fopen(s, "wt");
assert(ffw != NULL);
}
for(i=0; i<256; i++)
fprintf(ffw,"%f\t",Fw[i].real);
fprintf(ffw,"\n");
}
void dump_e(float e_hz[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fe == NULL) {
sprintf(s,"%s_e.txt", prefix);
fe = fopen(s, "wt");
assert(fe != NULL);
}
for(i=0; i<500/2; i++)
fprintf(fe,"%f\t",e_hz[i]);
fprintf(fe,"\n");
for(i=500/2; i<500; i++)
fprintf(fe,"%f\t",e_hz[i]);
fprintf(fe,"\n");
}
void dump_sq(float sq[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fsq == NULL) {
sprintf(s,"%s_sq.txt", prefix);
fsq = fopen(s, "wt");
assert(fsq != NULL);
}
for(i=0; i<M/2; i++)
fprintf(fsq,"%f\t",sq[i]);
fprintf(fsq,"\n");
for(i=M/2; i<M; i++)
fprintf(fsq,"%f\t",sq[i]);
fprintf(fsq,"\n");
}
void dump_dec(COMP Fw[]) {
int i;
char s[MAX_STR];
if (!dumpon) return;
if (fdec == NULL) {
sprintf(s,"%s_dec.txt", prefix);
fdec = fopen(s, "wt");
assert(fdec != NULL);
}
for(i=0; i<320/5; i++)
fprintf(fdec,"%f\t",Fw[i].real);
fprintf(fdec,"\n");
}
void dump_bg(float e, float bg_est, float percent_uv) {
char s[MAX_STR];
if (!dumpon) return;
if (fbg == NULL) {
sprintf(s,"%s_bg.txt", prefix);
fbg = fopen(s, "wt");
assert(fbg != NULL);
}
fprintf(fbg,"%f\t%f\t%f\n", e, bg_est, percent_uv);
}
void dump_E(float E) {
char s[MAX_STR];
if (!dumpon) return;
if (fE == NULL) {
sprintf(s,"%s_E.txt", prefix);
fE = fopen(s, "wt");
assert(fE != NULL);
}
fprintf(fE,"%f\n", 10.0*log10(E));
}

63
libs/libcodec2/src/dump.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: dump.h
AUTHOR......: David Rowe
DATE CREATED: 25/8/09
Routines to dump data to text files for Octave analysis.
\*---------------------------------------------------------------------------*/
/*
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __DUMP__
#define __DUMP__
void dump_on(char filename_prefix[]);
void dump_off();
void dump_Sn(float Sn[]);
void dump_Sw(COMP Sw[]);
void dump_Sw_(COMP Sw_[]);
/* amplitude modelling */
void dump_model(MODEL *m);
void dump_quantised_model(MODEL *m);
void dump_Pw(COMP Pw[]);
void dump_lsp(float lsp[]);
void dump_ak(float ak[], int order);
void dump_E(float E);
/* phase modelling */
void dump_snr(float snr);
void dump_phase(float phase[], int L);
void dump_phase_(float phase[], int L);
/* NLP states */
void dump_sq(float sq[]);
void dump_dec(COMP Fw[]);
void dump_Fw(COMP Fw[]);
void dump_e(float e_hz[]);
/* post filter */
void dump_bg(float e, float bg_est, float percent_uv);
#endif

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/*---------------------------------------------------------------------------*\
FILE........: four1.c
AUTHOR......: David Rowe
DATE CREATED: 20/2/95
Numerical Recipies in C FFT function. I have a nasty licence so please
replace me.
\*---------------------------------------------------------------------------*/
#include <math.h>
#define SWAP(a,b) tempr=(a);(a)=(b);(b)=tempr
void four1(data,nn,isign)
float data[];
int nn,isign;
{
int n,mmax,m,j,istep,i;
double wtemp,wr,wpr,wpi,wi,theta;
float tempr,tempi;
n=nn << 1;
j=1;
for (i=1;i<n;i+=2) {
if (j > i) {
SWAP(data[j],data[i]);
SWAP(data[j+1],data[i+1]);
}
m=n >> 1;
while (m >= 2 && j > m) {
j -= m;
m >>= 1;
}
j += m;
}
mmax=2;
while (n > mmax) {
istep=2*mmax;
theta=6.28318530717959/(isign*mmax);
wtemp=sin(0.5*theta);
wpr = -2.0*wtemp*wtemp;
wpi=sin(theta);
wr=1.0;
wi=0.0;
for (m=1;m<mmax;m+=2) {
for (i=m;i<=n;i+=istep) {
j=i+mmax;
tempr=wr*data[j]-wi*data[j+1];
tempi=wr*data[j+1]+wi*data[j];
data[j]=data[i]-tempr;
data[j+1]=data[i+1]-tempi;
data[i] += tempr;
data[i+1] += tempi;
}
wr=(wtemp=wr)*wpr-wi*wpi+wr;
wi=wi*wpr+wtemp*wpi+wi;
}
mmax=istep;
}
}
#undef SWAP

18
libs/libcodec2/src/four1.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: four1.h
AUTHOR......: David Rowe
DATE CREATED: 20/2/95
Numerical Recipies in C FFT function. I have a nasty licence so please
replace me.
\*---------------------------------------------------------------------------*/
#ifndef __FOUR1__
#define __FOUR1__
void four1(float x[], int n, int isign);
#endif /* __FOUR1__ */

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libs/libcodec2/src/fq20.sh Executable file
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#!/bin/sh
# fq20.shsh
# David Rowe 27 July 2010
#
# Decode a file with fully quantised codec at 20ms frame rate
../src/sinedec ../raw/$1.raw $1.mdl -o $1_phase0_lsp_20_EWo2.raw --phase 0 --lpc 10 --lsp --postfilter --dec

50
libs/libcodec2/src/globals.c Normal file
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/*---------------------------------------------------------------------------*\
FILE........: globals.c
AUTHOR......: David Rowe
DATE CREATED: 11/5/94
Globals for sinusoidal speech coder.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "sine.h" /* global defines for coder */
/* Globals used in encoder and decoder */
int frames; /* number of frames processed so far */
float Sn[M]; /* float input speech samples */
MODEL model; /* model parameters for the current frame */
int Nw; /* number of samples in analysis window */
float sig; /* energy of current frame */
/* Globals used in encoder */
float w[M]; /* time domain hamming window */
COMP W[FFT_ENC]; /* DFT of w[] */
COMP Sw[FFT_ENC]; /* DFT of current frame */
/* Globals used in decoder */
COMP Sw_[FFT_ENC]; /* DFT of all voiced synthesised signal */
float Sn_[AW_DEC]; /* synthesised speech */
float Pn[AW_DEC]; /* time domain Parzen (trapezoidal) window */

48
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/*---------------------------------------------------------------------------*\
FILE........: globals.h
AUTHOR......: David Rowe
DATE CREATED: 1/11/94
Globals for sinusoidal speech coder.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
/* Globals used in encoder and decoder */
extern int frames; /* number of frames processed so far */
extern float Sn[]; /* float input speech samples */
extern MODEL model; /* model parameters for the current frame */
extern int Nw; /* number of samples in analysis window */
extern float sig; /* energy of current frame */
/* Globals used in encoder */
extern float w[]; /* time domain hamming window */
extern COMP W[]; /* frequency domain hamming window */
extern COMP Sw[]; /* DFT of current frame */
extern COMP Sw_[]; /* DFT of all voiced synthesised signal */
/* Globals used in decoder */
extern float Sn_[]; /* output synthesised speech samples */
extern float Pn[]; /* time domain Parzen (trapezoidal) window */

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libs/libcodec2/src/interp.c Normal file
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/*---------------------------------------------------------------------------*\
FILE........: interp.c
AUTHOR......: David Rowe
DATE CREATED: 9/10/09
Interpolation of 20ms frames to 10ms frames.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <assert.h>
#include <math.h>
#include <string.h>
#include "defines.h"
#include "interp.h"
float sample_log_amp(MODEL *model, float w);
/*---------------------------------------------------------------------------*\
FUNCTION....: interp()
AUTHOR......: David Rowe
DATE CREATED: 22/8/10
Given two frames decribed by model parameters 20ms apart, determines
the model parameters of the 10ms frame between them. Assumes
voicing is available for middle (interpolated) frame. Outputs are
amplitudes and Wo for the interpolated frame.
This version can interpolate the amplitudes between two frames of
different Wo and L.
\*---------------------------------------------------------------------------*/
void interpolate(
MODEL *interp, /* interpolated model params */
MODEL *prev, /* previous frames model params */
MODEL *next /* next frames model params */
)
{
int l;
float w,log_amp;
/* Wo depends on voicing of this and adjacent frames */
if (interp->voiced) {
if (prev->voiced && next->voiced)
interp->Wo = (prev->Wo + next->Wo)/2.0;
if (!prev->voiced && next->voiced)
interp->Wo = next->Wo;
if (prev->voiced && !next->voiced)
interp->Wo = prev->Wo;
}
else {
interp->Wo = TWO_PI/P_MAX;
}
interp->L = PI/interp->Wo;
/* Interpolate amplitudes using linear interpolation in log domain */
for(l=1; l<=interp->L; l++) {
w = l*interp->Wo;
log_amp = (sample_log_amp(prev, w) + sample_log_amp(next, w))/2.0;
interp->A[l] = pow(10.0, log_amp);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: sample_log_amp()
AUTHOR......: David Rowe
DATE CREATED: 22/8/10
Samples the amplitude envelope at an arbitrary frequency w. Uses
linear interpolation in the log domain to sample between harmonic
amplitudes.
\*---------------------------------------------------------------------------*/
float sample_log_amp(MODEL *model, float w)
{
int m;
float f, log_amp;
assert(w > 0.0); assert (w <= PI);
m = floor(w/model->Wo + 0.5);
f = (w - m*model->Wo)/w;
assert(f <= 1.0);
if (m < 1) {
log_amp = f*log10(model->A[1]);
}
else if ((m+1) > model->L) {
log_amp = (1.0-f)*log10(model->A[model->L]);
}
else {
log_amp = (1.0-f)*log10(model->A[m]) + f*log10(model->A[m+1]);
}
return log_amp;
}

34
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/*---------------------------------------------------------------------------*\
FILE........: interp.h
AUTHOR......: David Rowe
DATE CREATED: 9/10/09
Interpolation of 20ms frames to 10ms frames.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __INTERP__
#define __INTERP__
void interpolate(MODEL *interp, MODEL *prev, MODEL *next);
#endif

9
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#!/bin/sh
# listensim.sh
# David Rowe 10 Sep 2009
#
# Listen to files processed with sim.sh
../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw $1_phase0_lsp.raw $2 $3

15
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#!/bin/sh
# listen1.sh
# David Rowe 10 Sep 2009
#
# Run menu with common sample file options, headphone version
#../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw ../raw/$1_g729a.raw $2 $3 -d /dev/dsp1
# compare to other codecs
#../script/menu.sh ../raw/$1.raw $1_phase0_lsp.raw $1_phase0_lsp_20.raw ../raw/$1_g729a.raw ../raw/$1_gsm13k.raw ../raw/$1_speex_8k.raw $2 $3 -d /dev/dsp1
../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_test.raw ../raw/$1_g729a.raw $2 $3 -d /dev/dsp1

9
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#!/bin/sh
# listensim.sh
# David Rowe 10 Sep 2009
#
# Listen to files processed with sim.sh
../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw $1_phase0_lsp_dec.raw $2 $3

253
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/*---------------------------------------------------------------------------*\
FILE........: lpc.c
AUTHOR......: David Rowe
DATE CREATED: 30/9/90
Linear Prediction functions written in C.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#define LPC_MAX_N 512 /* maximum no. of samples in frame */
#define PI 3.141592654 /* mathematical constant */
#include <assert.h>
#include <math.h>
#include "defines.h"
#include "lpc.h"
/*---------------------------------------------------------------------------*\
hanning_window()
Hanning windows a frame of speech samples.
\*---------------------------------------------------------------------------*/
void hanning_window(
float Sn[], /* input frame of speech samples */
float Wn[], /* output frame of windowed samples */
int Nsam /* number of samples */
)
{
int i; /* loop variable */
for(i=0; i<Nsam; i++)
Wn[i] = Sn[i]*(0.5 - 0.5*cos(2*PI*(float)i/(Nsam-1)));
}
/*---------------------------------------------------------------------------*\
autocorrelate()
Finds the first P autocorrelation values of an array of windowed speech
samples Sn[].
\*---------------------------------------------------------------------------*/
void autocorrelate(
float Sn[], /* frame of Nsam windowed speech samples */
float Rn[], /* array of P+1 autocorrelation coefficients */
int Nsam, /* number of windowed samples to use */
int order /* order of LPC analysis */
)
{
int i,j; /* loop variables */
for(j=0; j<order+1; j++) {
Rn[j] = 0.0;
for(i=0; i<Nsam-j; i++)
Rn[j] += Sn[i]*Sn[i+j];
}
}
/*---------------------------------------------------------------------------*\
levinson_durbin()
Given P+1 autocorrelation coefficients, finds P Linear Prediction Coeff.
(LPCs) where P is the order of the LPC all-pole model. The Levinson-Durbin
algorithm is used, and is described in:
J. Makhoul
"Linear prediction, a tutorial review"
Proceedings of the IEEE
Vol-63, No. 4, April 1975
\*---------------------------------------------------------------------------*/
void levinson_durbin(
float R[], /* order+1 autocorrelation coeff */
float lpcs[], /* order+1 LPC's */
int order /* order of the LPC analysis */
)
{
float E[LPC_MAX+1];
float k[LPC_MAX+1];
float a[LPC_MAX+1][LPC_MAX+1];
float sum;
int i,j; /* loop variables */
E[0] = R[0]; /* Equation 38a, Makhoul */
for(i=1; i<=order; i++) {
sum = 0.0;
for(j=1; j<=i-1; j++)
sum += a[i-1][j]*R[i-j];
k[i] = -1.0*(R[i] + sum)/E[i-1]; /* Equation 38b, Makhoul */
if (fabs(k[i]) > 1.0)
k[i] = 0.0;
a[i][i] = k[i];
for(j=1; j<=i-1; j++)
a[i][j] = a[i-1][j] + k[i]*a[i-1][i-j]; /* Equation 38c, Makhoul */
E[i] = (1-k[i]*k[i])*E[i-1]; /* Equation 38d, Makhoul */
}
for(i=1; i<=order; i++)
lpcs[i] = a[order][i];
lpcs[0] = 1.0;
}
/*---------------------------------------------------------------------------*\
inverse_filter()
Inverse Filter, A(z). Produces an array of residual samples from an array
of input samples and linear prediction coefficients.
The filter memory is stored in the first order samples of the input array.
\*---------------------------------------------------------------------------*/
void inverse_filter(
float Sn[], /* Nsam input samples */
float a[], /* LPCs for this frame of samples */
int Nsam, /* number of samples */
float res[], /* Nsam residual samples */
int order /* order of LPC */
)
{
int i,j; /* loop variables */
for(i=0; i<Nsam; i++) {
res[i] = 0.0;
for(j=0; j<=order; j++)
res[i] += Sn[i-j]*a[j];
}
}
/*---------------------------------------------------------------------------*\
synthesis_filter()
C version of the Speech Synthesis Filter, 1/A(z). Given an array of
residual or excitation samples, and the the LP filter coefficients, this
function will produce an array of speech samples. This filter structure is
IIR.
The synthesis filter has memory as well, this is treated in the same way
as the memory for the inverse filter (see inverse_filter() notes above).
The difference is that the memory for the synthesis filter is stored in
the output array, wheras the memory of the inverse filter is stored in the
input array.
Note: the calling function must update the filter memory.
\*---------------------------------------------------------------------------*/
void synthesis_filter(
float res[], /* Nsam input residual (excitation) samples */
float a[], /* LPCs for this frame of speech samples */
int Nsam, /* number of speech samples */
int order, /* LPC order */
float Sn_[] /* Nsam output synthesised speech samples */
)
{
int i,j; /* loop variables */
/* Filter Nsam samples */
for(i=0; i<Nsam; i++) {
Sn_[i] = res[i]*a[0];
for(j=1; j<=order; j++)
Sn_[i] -= Sn_[i-j]*a[j];
}
}
/*---------------------------------------------------------------------------*\
find_aks()
This function takes a frame of samples, and determines the linear
prediction coefficients for that frame of samples.
\*---------------------------------------------------------------------------*/
void find_aks(
float Sn[], /* Nsam samples with order sample memory */
float a[], /* order+1 LPCs with first coeff 1.0 */
int Nsam, /* number of input speech samples */
int order, /* order of the LPC analysis */
float *E /* residual energy */
)
{
float Wn[LPC_MAX_N]; /* windowed frame of Nsam speech samples */
float R[LPC_MAX+1]; /* order+1 autocorrelation values of Sn[] */
int i;
assert(order < LPC_MAX);
assert(Nsam < LPC_MAX_N);
hanning_window(Sn,Wn,Nsam);
autocorrelate(Wn,R,Nsam,order);
levinson_durbin(R,a,order);
*E = 0.0;
for(i=0; i<=order; i++)
*E += a[i]*R[i];
if (*E < 0.0)
*E = 1E-12;
}
/*---------------------------------------------------------------------------*\
weight()
Weights a vector of LPCs.
\*---------------------------------------------------------------------------*/
void weight(
float ak[], /* vector of order+1 LPCs */
float gamma, /* weighting factor */
int order, /* num LPCs (excluding leading 1.0) */
float akw[] /* weighted vector of order+1 LPCs */
)
{
int i;
for(i=1; i<=order; i++)
akw[i] = ak[i]*pow(gamma,(float)i);
}

42
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/*---------------------------------------------------------------------------*\
FILE........: lpc.h
AUTHOR......: David Rowe
DATE CREATED: 24/8/09
Linear Prediction functions written in C.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __LPC__
#define __LPC__
#define LPC_MAX_ORDER 20
void hanning_window(float Sn[], float Wn[], int Nsam);
void autocorrelate(float Sn[], float Rn[], int Nsam, int order);
void levinson_durbin(float R[], float lpcs[], int order);
void inverse_filter(float Sn[], float a[], int Nsam, float res[], int order);
void synthesis_filter(float res[], float a[], int Nsam, int order, float Sn_[]);
void find_aks(float Sn[], float a[], int Nsam, int order, float *E);
void weight(float ak[], float gamma, int order, float akw[]);
#endif

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/*---------------------------------------------------------------------------*\
FILE........: lsp.c
AUTHOR......: David Rowe
DATE CREATED: 24/2/93
This file contains functions for LPC to LSP conversion and LSP to
LPC conversion. Note that the LSP coefficients are not in radians
format but in the x domain of the unit circle.
\*---------------------------------------------------------------------------*/
#include "defines.h"
#include "lsp.h"
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
/*---------------------------------------------------------------------------*\
Introduction to Line Spectrum Pairs (LSPs)
------------------------------------------
LSPs are used to encode the LPC filter coefficients {ak} for
transmission over the channel. LSPs have several properties (like
less sensitivity to quantisation noise) that make them superior to
direct quantisation of {ak}.
A(z) is a polynomial of order lpcrdr with {ak} as the coefficients.
A(z) is transformed to P(z) and Q(z) (using a substitution and some
algebra), to obtain something like:
A(z) = 0.5[P(z)(z+z^-1) + Q(z)(z-z^-1)] (1)
As you can imagine A(z) has complex zeros all over the z-plane. P(z)
and Q(z) have the very neat property of only having zeros _on_ the
unit circle. So to find them we take a test point z=exp(jw) and
evaluate P (exp(jw)) and Q(exp(jw)) using a grid of points between 0
and pi.
The zeros (roots) of P(z) also happen to alternate, which is why we
swap coefficients as we find roots. So the process of finding the
LSP frequencies is basically finding the roots of 5th order
polynomials.
The root so P(z) and Q(z) occur in symmetrical pairs at +/-w, hence
the name Line Spectrum Pairs (LSPs).
To convert back to ak we just evaluate (1), "clocking" an impulse
thru it lpcrdr times gives us the impulse response of A(z) which is
{ak}.
\*---------------------------------------------------------------------------*/
/*---------------------------------------------------------------------------*\
FUNCTION....: cheb_poly_eva()
AUTHOR......: David Rowe
DATE CREATED: 24/2/93
This function evalutes a series of chebyshev polynomials
FIXME: performing memory allocation at run time is very inefficient,
replace with stack variables of MAX_P size.
\*---------------------------------------------------------------------------*/
float cheb_poly_eva(float *coef,float x,int m)
/* float coef[] coefficients of the polynomial to be evaluated */
/* float x the point where polynomial is to be evaluated */
/* int m order of the polynomial */
{
int i;
float *T,*t,*u,*v,sum;
/* Allocate memory for chebyshev series formulation */
if((T = (float *)malloc((m/2+1)*sizeof(float))) == NULL){
fprintf(stderr, "not enough memory to allocate buffer\n");
exit(1);
}
/* Initialise pointers */
t = T; /* T[i-2] */
*t++ = 1.0;
u = t--; /* T[i-1] */
*u++ = x;
v = u--; /* T[i] */
/* Evaluate chebyshev series formulation using iterative approach */
for(i=2;i<=m/2;i++)
*v++ = (2*x)*(*u++) - *t++; /* T[i] = 2*x*T[i-1] - T[i-2] */
sum=0.0; /* initialise sum to zero */
t = T; /* reset pointer */
/* Evaluate polynomial and return value also free memory space */
for(i=0;i<=m/2;i++)
sum+=coef[(m/2)-i]**t++;
free(T);
return sum;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: lpc_to_lsp()
AUTHOR......: David Rowe
DATE CREATED: 24/2/93
This function converts LPC coefficients to LSP coefficients.
\*---------------------------------------------------------------------------*/
int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta)
/* float *a lpc coefficients */
/* int lpcrdr order of LPC coefficients (10) */
/* float *freq LSP frequencies in radians */
/* int nb number of sub-intervals (4) */
/* float delta grid spacing interval (0.02) */
{
float psuml,psumr,psumm,temp_xr,xl,xr,xm;
float temp_psumr;
int i,j,m,flag,k;
float *Q; /* ptrs for memory allocation */
float *P;
float *px; /* ptrs of respective P'(z) & Q'(z) */
float *qx;
float *p;
float *q;
float *pt; /* ptr used for cheb_poly_eval()
whether P' or Q' */
int roots=0; /* number of roots found */
flag = 1;
m = lpcrdr/2; /* order of P'(z) & Q'(z) polynimials */
/* Allocate memory space for polynomials */
Q = (float *) malloc((m+1)*sizeof(float));
P = (float *) malloc((m+1)*sizeof(float));
if( (P == NULL) || (Q == NULL) ) {
fprintf(stderr,"not enough memory to allocate buffer\n");
exit(1);
}
/* determine P'(z)'s and Q'(z)'s coefficients where
P'(z) = P(z)/(1 + z^(-1)) and Q'(z) = Q(z)/(1-z^(-1)) */
px = P; /* initilaise ptrs */
qx = Q;
p = px;
q = qx;
*px++ = 1.0;
*qx++ = 1.0;
for(i=1;i<=m;i++){
*px++ = a[i]+a[lpcrdr+1-i]-*p++;
*qx++ = a[i]-a[lpcrdr+1-i]+*q++;
}
px = P;
qx = Q;
for(i=0;i<m;i++){
*px = 2**px;
*qx = 2**qx;
px++;
qx++;
}
px = P; /* re-initialise ptrs */
qx = Q;
/* Search for a zero in P'(z) polynomial first and then alternate to Q'(z).
Keep alternating between the two polynomials as each zero is found */
xr = 0; /* initialise xr to zero */
xl = 1.0; /* start at point xl = 1 */
for(j=0;j<lpcrdr;j++){
if(j%2) /* determines whether P' or Q' is eval. */
pt = qx;
else
pt = px;
psuml = cheb_poly_eva(pt,xl,lpcrdr); /* evals poly. at xl */
flag = 1;
while(flag && (xr >= -1.0)){
xr = xl - delta ; /* interval spacing */
psumr = cheb_poly_eva(pt,xr,lpcrdr);/* poly(xl-delta_x) */
temp_psumr = psumr;
temp_xr = xr;
/* if no sign change increment xr and re-evaluate
poly(xr). Repeat til sign change. if a sign change has
occurred the interval is bisected and then checked again
for a sign change which determines in which interval the
zero lies in. If there is no sign change between poly(xm)
and poly(xl) set interval between xm and xr else set
interval between xl and xr and repeat till root is located
within the specified limits */
if((psumr*psuml)<0.0){
roots++;
psumm=psuml;
for(k=0;k<=nb;k++){
xm = (xl+xr)/2; /* bisect the interval */
psumm=cheb_poly_eva(pt,xm,lpcrdr);
if(psumm*psuml>0.){
psuml=psumm;
xl=xm;
}
else{
psumr=psumm;
xr=xm;
}
}
/* once zero is found, reset initial interval to xr */
freq[j] = (xm);
xl = xm;
flag = 0; /* reset flag for next search */
}
else{
psuml=temp_psumr;
xl=temp_xr;
}
}
}
free(P); /* free memory space */
free(Q);
/* convert from x domain to radians */
for(i=0; i<lpcrdr; i++) {
freq[i] = acos(freq[i]);
}
return(roots);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: lsp_to_lpc()
AUTHOR......: David Rowe
DATE CREATED: 24/2/93
This function converts LSP coefficients to LPC coefficients. In the
Speex code we worked out a wayto simplify this significantly.
\*---------------------------------------------------------------------------*/
void lsp_to_lpc(float *freq, float *ak, int lpcrdr)
/* float *freq array of LSP frequencies in radians */
/* float *ak array of LPC coefficients */
/* int lpcrdr order of LPC coefficients */
{
int i,j;
float xout1,xout2,xin1,xin2;
float *Wp;
float *pw,*n1,*n2,*n3,*n4;
int m = lpcrdr/2;
/* convert from radians to the x=cos(w) domain */
for(i=0; i<lpcrdr; i++)
freq[i] = cos(freq[i]);
if((Wp = (float *) malloc((4*m+2)*sizeof(float))) == NULL){
printf("not enough memory to allocate buffer\n");
exit(1);
}
pw = Wp;
/* initialise contents of array */
for(i=0;i<=4*m+1;i++){ /* set contents of buffer to 0 */
*pw++ = 0.0;
}
/* Set pointers up */
pw = Wp;
xin1 = 1.0;
xin2 = 1.0;
/* reconstruct P(z) and Q(z) by cascading second order polynomials
in form 1 - 2xz(-1) +z(-2), where x is the LSP coefficient */
for(j=0;j<=lpcrdr;j++){
for(i=0;i<m;i++){
n1 = pw+(i*4);
n2 = n1 + 1;
n3 = n2 + 1;
n4 = n3 + 1;
xout1 = xin1 - 2*(freq[2*i]) * *n1 + *n2;
xout2 = xin2 - 2*(freq[2*i+1]) * *n3 + *n4;
*n2 = *n1;
*n4 = *n3;
*n1 = xin1;
*n3 = xin2;
xin1 = xout1;
xin2 = xout2;
}
xout1 = xin1 + *(n4+1);
xout2 = xin2 - *(n4+2);
ak[j] = (xout1 + xout2)*0.5;
*(n4+1) = xin1;
*(n4+2) = xin2;
xin1 = 0.0;
xin2 = 0.0;
}
free(Wp);
}

20
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/*---------------------------------------------------------------------------*\
FILE........: lsp.c
AUTHOR......: David Rowe
DATE CREATED: 24/2/93
This file contains functions for LPC to LSP conversion and LSP to
LPC conversion. Note that the LSP coefficients are not in radians
format but in the x domain of the unit circle.
\*---------------------------------------------------------------------------*/
#ifndef __LSP__
#define __LSP__
int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta);
void lsp_to_lpc(float *freq, float *ak, int lpcrdr);
#endif

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/*---------------------------------------------------------------------------*\
FILE........: nlp.c
AUTHOR......: David Rowe
DATE CREATED: 23/3/93
Non Linear Pitch (NLP) estimation functions.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "defines.h"
#include "nlp.h"
#include "dump.h"
#include "four1.h"
#include <assert.h>
#include <math.h>
#include <stdlib.h>
/*---------------------------------------------------------------------------*\
DEFINES
\*---------------------------------------------------------------------------*/
#define PMAX_M 600 /* maximum NLP analysis window size */
#define COEFF 0.95 /* notch filter parameter */
#define PE_FFT_SIZE 512 /* DFT size for pitch estimation */
#define DEC 5 /* decimation factor */
#define SAMPLE_RATE 8000
#define PI 3.141592654 /* mathematical constant */
#define T 0.1 /* threshold for local minima candidate */
#define F0_MAX 500
#define CNLP 0.3 /* post processor constant */
#define NLP_NTAP 48 /* Decimation LPF order */
/*---------------------------------------------------------------------------*\
GLOBALS
\*---------------------------------------------------------------------------*/
/* 48 tap 600Hz low pass FIR filter coefficients */
float nlp_fir[] = {
-1.0818124e-03,
-1.1008344e-03,
-9.2768838e-04,
-4.2289438e-04,
5.5034190e-04,
2.0029849e-03,
3.7058509e-03,
5.1449415e-03,
5.5924666e-03,
4.3036754e-03,
8.0284511e-04,
-4.8204610e-03,
-1.1705810e-02,
-1.8199275e-02,
-2.2065282e-02,
-2.0920610e-02,
-1.2808831e-02,
3.2204775e-03,
2.6683811e-02,
5.5520624e-02,
8.6305944e-02,
1.1480192e-01,
1.3674206e-01,
1.4867556e-01,
1.4867556e-01,
1.3674206e-01,
1.1480192e-01,
8.6305944e-02,
5.5520624e-02,
2.6683811e-02,
3.2204775e-03,
-1.2808831e-02,
-2.0920610e-02,
-2.2065282e-02,
-1.8199275e-02,
-1.1705810e-02,
-4.8204610e-03,
8.0284511e-04,
4.3036754e-03,
5.5924666e-03,
5.1449415e-03,
3.7058509e-03,
2.0029849e-03,
5.5034190e-04,
-4.2289438e-04,
-9.2768838e-04,
-1.1008344e-03,
-1.0818124e-03
};
typedef struct {
float sq[PMAX_M]; /* squared speech samples */
float mem_x,mem_y; /* memory for notch filter */
float mem_fir[NLP_NTAP]; /* decimation FIR filter memory */
} NLP;
float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax);
float post_process_sub_multiples(COMP Fw[],
int pmin, int pmax, float gmax, int gmax_bin,
float *prev_Wo);
/*---------------------------------------------------------------------------*\
nlp_create()
Initialisation function for NLP pitch estimator.
\*---------------------------------------------------------------------------*/
void *nlp_create()
{
NLP *nlp;
int i;
nlp = (NLP*)malloc(sizeof(NLP));
if (nlp == NULL)
return NULL;
for(i=0; i<PMAX_M; i++)
nlp->sq[i] = 0.0;
nlp->mem_x = 0.0;
nlp->mem_y = 0.0;
for(i=0; i<NLP_NTAP; i++)
nlp->mem_fir[i] = 0.0;
return (void*)nlp;
}
/*---------------------------------------------------------------------------*\
nlp_destory()
Initialisation function for NLP pitch estimator.
\*---------------------------------------------------------------------------*/
void nlp_destroy(void *nlp_state)
{
assert(nlp_state != NULL);
free(nlp_state);
}
/*---------------------------------------------------------------------------*\
nlp()
Determines the pitch in samples using the Non Linear Pitch (NLP)
algorithm [1]. Returns the fundamental in Hz. Note that the actual
pitch estimate is for the centre of the M sample Sn[] vector, not
the current N sample input vector. This is (I think) a delay of 2.5
frames with N=80 samples. You should align further analysis using
this pitch estimate to be centred on the middle of Sn[].
Two post processors have been tried, the MBE version (as discussed
in [1]), and a post processor that checks sub-multiples. Both
suffer occasional gross pitch errors (i.e. neither are perfect). In
the presence of background noise the sub-multiple algorithm tends
towards low F0 which leads to better sounding background noise than
the MBE post processor.
A good way to test and develop the NLP pitch estimator is using the
tnlp (codec2/unittest) and the codec2/octave/plnlp.m Octave script.
A pitch tracker searching a few frames forward and backward in time
would be a useful addition.
References:
[1] http://www.itr.unisa.edu.au/~steven/thesis/dgr.pdf Chapter 4
\*---------------------------------------------------------------------------*/
float nlp(
void *nlp_state,
float Sn[], /* input speech vector */
int n, /* frames shift (no. new samples in Sn[]) */
int m, /* analysis window size */
int pmin, /* minimum pitch value */
int pmax, /* maximum pitch value */
float *pitch, /* estimated pitch period in samples */
COMP Sw[], /* Freq domain version of Sn[] */
float *prev_Wo
)
{
NLP *nlp;
float notch; /* current notch filter output */
COMP Fw[PE_FFT_SIZE]; /* DFT of squared signal */
float gmax;
int gmax_bin;
int i,j;
float best_f0;
assert(nlp_state != NULL);
nlp = (NLP*)nlp_state;
/* Square, notch filter at DC, and LP filter vector */
for(i=m-n; i<M; i++) /* square latest speech samples */
nlp->sq[i] = Sn[i]*Sn[i];
for(i=m-n; i<m; i++) { /* notch filter at DC */
notch = nlp->sq[i] - nlp->mem_x;
notch += COEFF*nlp->mem_y;
nlp->mem_x = nlp->sq[i];
nlp->mem_y = notch;
nlp->sq[i] = notch;
}
for(i=m-n; i<m; i++) { /* FIR filter vector */
for(j=0; j<NLP_NTAP-1; j++)
nlp->mem_fir[j] = nlp->mem_fir[j+1];
nlp->mem_fir[NLP_NTAP-1] = nlp->sq[i];
nlp->sq[i] = 0.0;
for(j=0; j<NLP_NTAP; j++)
nlp->sq[i] += nlp->mem_fir[j]*nlp_fir[j];
}
/* Decimate and DFT */
for(i=0; i<PE_FFT_SIZE; i++) {
Fw[i].real = 0.0;
Fw[i].imag = 0.0;
}
for(i=0; i<m/DEC; i++) {
Fw[i].real = nlp->sq[i*DEC]*(0.5 - 0.5*cos(2*PI*i/(m/DEC-1)));
}
dump_dec(Fw);
four1(&Fw[-1].imag,PE_FFT_SIZE,1);
for(i=0; i<PE_FFT_SIZE; i++)
Fw[i].real = Fw[i].real*Fw[i].real + Fw[i].imag*Fw[i].imag;
dump_sq(nlp->sq);
dump_Fw(Fw);
/* find global peak */
gmax = 0.0;
gmax_bin = PE_FFT_SIZE*DEC/pmax;
for(i=PE_FFT_SIZE*DEC/pmax; i<=PE_FFT_SIZE*DEC/pmin; i++) {
if (Fw[i].real > gmax) {
gmax = Fw[i].real;
gmax_bin = i;
}
}
best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin,
prev_Wo);
/* Shift samples in buffer to make room for new samples */
for(i=0; i<m-n; i++)
nlp->sq[i] = nlp->sq[i+n];
/* return pitch and F0 estimate */
*pitch = (float)SAMPLE_RATE/best_f0;
return(best_f0);
}
/*---------------------------------------------------------------------------*\
post_process_sub_multiples()
Given the global maximma of Fw[] we search interger submultiples for
local maxima. If local maxima exist and they are above an
experimentally derived threshold (OK a magic number I pulled out of
the air) we choose the submultiple as the F0 estimate.
The rational for this is that the lowest frequency peak of Fw[]
should be F0, as Fw[] can be considered the autocorrelation function
of Sw[] (the speech spectrum). However sometimes due to phase
effects the lowest frequency maxima may not be the global maxima.
This works OK in practice and favours low F0 values in the presence
of background noise which means the sinusoidal codec does an OK job
of synthesising the background noise. High F0 in background noise
tends to sound more periodic introducing annoying artifacts.
\*---------------------------------------------------------------------------*/
float post_process_sub_multiples(COMP Fw[],
int pmin, int pmax, float gmax, int gmax_bin,
float *prev_Wo)
{
int min_bin, cmax_bin;
int mult;
float thresh, best_f0;
int b, bmin, bmax, lmax_bin;
float lmax, cmax;
int prev_f0_bin;
/* post process estimate by searching submultiples */
mult = 2;
min_bin = PE_FFT_SIZE*DEC/pmax;
cmax_bin = gmax_bin;
prev_f0_bin = *prev_Wo*(4000.0/PI)*(PE_FFT_SIZE*DEC)/SAMPLE_RATE;
while(gmax_bin/mult >= min_bin) {
b = gmax_bin/mult; /* determine search interval */
bmin = 0.8*b;
bmax = 1.2*b;
if (bmin < min_bin)
bmin = min_bin;
/* lower threshold to favour previous frames pitch estimate,
this is a form of pitch tracking */
if ((prev_f0_bin > bmin) && (prev_f0_bin < bmax))
thresh = CNLP*0.5*gmax;
else
thresh = CNLP*gmax;
lmax = 0;
lmax_bin = bmin;
for (b=bmin; b<=bmax; b++) /* look for maximum in interval */
if (Fw[b].real > lmax) {
lmax = Fw[b].real;
lmax_bin = b;
}
if (lmax > thresh)
if ((lmax > Fw[lmax_bin-1].real) && (lmax > Fw[lmax_bin+1].real)) {
cmax = lmax;
cmax_bin = lmax_bin;
}
mult++;
}
best_f0 = (float)cmax_bin*SAMPLE_RATE/(PE_FFT_SIZE*DEC);
return best_f0;
}

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/*---------------------------------------------------------------------------*\
FILE........: nlp.c
AUTHOR......: David Rowe
DATE CREATED: 23/3/93
Non Linear Pitch (NLP) estimation functions.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __NLP__
#define __NLP__
void *nlp_create();
void nlp_destroy(void *nlp_state);
float nlp(void *nlp_state, float Sn[], int n, int m, int pmin, int pmax,
float *pitch, COMP Sw[], float *prev_Wo);
float test_candidate_mbe(COMP Sw[], float f0, COMP Sw_[]);
#endif

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/*
Copyright (C) 2010 Perens LLC <bruce@perens.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
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 General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include "defines.h"
#include "quantise.h"
#include <stdio.h>
/* Compile-time constants */
/* Size of unsigned char in bits. Assumes 8 bits-per-char. */
static const unsigned int WordSize = 8;
/* Mask to pick the bit component out of bitIndex. */
static const unsigned int IndexMask = 0x7;
/* Used to pick the word component out of bitIndex. */
static const unsigned int ShiftRight = 3;
/** Pack a bit field into a bit string, encoding the field in Gray code.
*
* The output is an array of unsigned char data. The fields are efficiently
* packed into the bit string. The Gray coding is a naive attempt to reduce
* the effect of single-bit errors, we expect to do a better job as the
* codec develops.
*
* This code would be simpler if it just set one bit at a time in the string,
* but would hit the same cache line more often. I'm not sure the complexity
* gains us anything here.
*
* Although field is currently of int type rather than unsigned for
* compatibility with the rest of the code, indices are always expected to
* be >= 0.
*/
void
pack(
unsigned char * bitArray, /* The output bit string. */
unsigned int * bitIndex, /* Index into the string in BITS, not bytes.*/
int field, /* The bit field to be packed. */
unsigned int fieldWidth/* Width of the field in BITS, not bytes. */
)
{
/* Convert the field to Gray code */
field = (field >> 1) ^ field;
do {
unsigned int bI = *bitIndex;
unsigned int bitsLeft = WordSize - (bI & IndexMask);
unsigned int sliceWidth =
bitsLeft < fieldWidth ? bitsLeft : fieldWidth;
unsigned int wordIndex = bI >> ShiftRight;
bitArray[wordIndex] |=
((unsigned char)((field >> (fieldWidth - sliceWidth))
<< (bitsLeft - sliceWidth)));
*bitIndex = bI + sliceWidth;
fieldWidth -= sliceWidth;
} while ( fieldWidth != 0 );
}
/** Unpack a field from a bit string, converting from Gray code to binary.
*
*/
int
unpack(
const unsigned char * bitArray, /* The input bit string. */
unsigned int * bitIndex, /* Index into the string in BITS, not bytes.*/
unsigned int fieldWidth/* Width of the field in BITS, not bytes. */
)
{
unsigned int field = 0;
do {
unsigned int bI = *bitIndex;
unsigned int bitsLeft = WordSize - (bI & IndexMask);
unsigned int sliceWidth =
bitsLeft < fieldWidth ? bitsLeft : fieldWidth;
field |= (((bitArray[bI >> ShiftRight] >> (bitsLeft - sliceWidth)) & ((1 << sliceWidth) - 1)) << (fieldWidth - sliceWidth));
*bitIndex = bI + sliceWidth;
fieldWidth -= sliceWidth;
} while ( fieldWidth != 0 );
/* Convert from Gray code to binary. Works for maximum 8-bit fields. */
unsigned int t = field ^ (field >> 8);
t ^= (t >> 4);
t ^= (t >> 2);
t ^= (t >> 1);
return t;
}

254
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/*---------------------------------------------------------------------------*\
FILE........: phase.c
AUTHOR......: David Rowe
DATE CREATED: 1/2/09
Functions for modelling and synthesising phase.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "defines.h"
#include "phase.h"
#include "four1.h"
#include <assert.h>
#include <math.h>
#include <string.h>
#include <stdlib.h>
#define VTHRESH 4.0
/*---------------------------------------------------------------------------*\
aks_to_H()
Samples the complex LPC synthesis filter spectrum at the harmonic
frequencies.
\*---------------------------------------------------------------------------*/
void aks_to_H(
MODEL *model, /* model parameters */
float aks[], /* LPC's */
float G, /* energy term */
COMP H[], /* complex LPC spectral samples */
int order
)
{
COMP Pw[FFT_DEC]; /* 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 */
int b; /* centre bin of harmonic */
float phi_; /* phase of LPC spectra */
r = TWO_PI/(FFT_DEC);
/* Determine DFT of A(exp(jw)) ------------------------------------------*/
for(i=0; i<FFT_DEC; i++) {
Pw[i].real = 0.0;
Pw[i].imag = 0.0;
}
for(i=0; i<=order; i++)
Pw[i].real = aks[i];
four1(&Pw[-1].imag,FFT_DEC,-1);
/* Sample magnitude and phase at harmonics */
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);
b = floor(m*model->Wo/r + 0.5);
Em = 0.0;
for(i=am; i<bm; i++)
Em += G/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
Am = sqrt(fabs(Em/(bm-am)));
phi_ = -atan2(Pw[b].imag,Pw[b].real);
H[m].real = Am*cos(phi_);
H[m].imag = Am*sin(phi_);
}
}
/*---------------------------------------------------------------------------*\
phase_synth_zero_order()
Synthesises phases based on SNR and a rule based approach. No phase
parameters are required apart from the SNR (which can be reduced to a
1 bit V/UV decision per frame).
The phase of each harmonic is modelled as the phase of a LPC
synthesis filter excited by an impulse. Unlike the first order
model the position of the impulse is not transmitted, so we create
an excitation pulse train using a rule based approach.
Consider a pulse train with a pulse starting time n=0, with pulses
repeated at a rate of Wo, the fundamental frequency. A pulse train
in the time domain is equivalent to harmonics in the frequency
domain. We can make an excitation pulse train using a sum of
sinsusoids:
for(m=1; m<=L; m++)
ex[n] = cos(m*Wo*n)
Note: the Octave script ../octave/phase.m is an example of this if
you would like to try making a pulse train.
The phase of each excitation harmonic is:
arg(E[m]) = mWo
where E[m] are the complex excitation (freq domain) samples,
arg(x), just returns the phase of a complex sample x.
As we don't transmit the pulse position for this model, we need to
synthesise it. Now the excitation pulses occur at a rate of Wo.
This means the phase of the first harmonic advances by N samples
over a synthesis frame of N samples. For example if Wo is pi/20
(200 Hz), then over a 10ms frame (N=80 samples), the phase of the
first harmonic would advance (pi/20)*80 = 4*pi or two complete
cycles.
We generate the excitation phase of the fundamental (first
harmonic):
arg[E[1]] = Wo*N;
We then relate the phase of the m-th excitation harmonic to the
phase of the fundamental as:
arg(E[m]) = m*arg(E[1])
This E[m] then gets passed through the LPC synthesis filter to
determine the final harmonic phase.
For a while there were prolems with low pitched males like hts1
sounding "clicky". The synthesied time domain waveform also looked
clicky. Many methods were tried to improve the sounds quality of
low pitched males. Finally adding a small amount of jitter to each
harmonic worked.
The current result sounds very close to the original phases, with
only 1 voicing bit per frame. For example hts1a using original
amplitudes and this phase model produces speech hard to distinguish
from speech synthesise with the orginal phases. The sound quality
of this patrtiallyuantised codec (nb original amplitudes) is higher
than g729, even though all the phase information has been
discarded.
NOTES:
1/ This synthesis model is effectvely the same as simple LPC-10
vocoders, and yet sounds much better. Why?
2/ I am pretty sure the Lincoln Lab sinusoidal coding guys (like xMBE
also from MIT) first described this zero phase model, I need to look
up the paper.
3/ Note that this approach could cause some discontinuities in
the phase at the edge of synthesis frames, as no attempt is made
to make sure that the phase tracks are continuous (the excitation
phases are continuous, but not the final phases after filtering
by the LPC spectra). Technically this is a bad thing. However
this may actually be a good thing, disturbing the phase tracks a
bit. More research needed, e.g. test a synthesis model that adds
a small delta-W to make phase tracks line up for voiced
harmonics.
4/ Why does this sound so great with 1 V/UV decision? Conventional
wisdom says mixed voicing is required for high qaulity speech.
\*---------------------------------------------------------------------------*/
void phase_synth_zero_order(
MODEL *model,
float aks[],
float *ex_phase /* excitation phase of fundamental */
)
{
int m;
float new_phi;
COMP Ex[MAX_AMP]; /* excitation samples */
COMP A_[MAX_AMP]; /* synthesised harmonic samples */
COMP H[MAX_AMP]; /* LPC freq domain samples */
float G;
float jitter;
G = 1.0;
aks_to_H(model,aks,G,H,LPC_ORD);
/*
Update excitation fundamental phase track, this sets the position
of each pitch pulse during voiced speech. After much experiment
I found that using just this frame Wo improved quality for UV
sounds compared to interpolating two frames Wo like this:
ex_phase[0] += (*prev_Wo+mode->Wo)*N/2;
*/
ex_phase[0] += (model->Wo)*N;
ex_phase[0] -= TWO_PI*floor(ex_phase[0]/TWO_PI + 0.5);
for(m=1; m<=model->L; m++) {
/* generate excitation */
if (model->voiced) {
/* This method of adding jitter really helped remove the clicky
sound in low pitched makes like hts1a. This moves the onset
of each harmonic over at +/- 0.25 of a sample.
*/
jitter = 0.25*(1.0 - 2.0*rand()/RAND_MAX);
Ex[m].real = cos(ex_phase[0]*m - jitter*model->Wo*m);
Ex[m].imag = sin(ex_phase[0]*m - jitter*model->Wo*m);
}
else {
/* When a few samples were tested I found that LPC filter
phase is not needed in the unvoiced case, but no harm in
keeping it.
*/
float phi = TWO_PI*(float)rand()/RAND_MAX;
Ex[m].real = cos(phi);
Ex[m].imag = sin(phi);
}
/* filter using LPC filter */
A_[m].real = H[m].real*Ex[m].real - H[m].imag*Ex[m].imag;
A_[m].imag = H[m].imag*Ex[m].real + H[m].real*Ex[m].imag;
/* modify sinusoidal phase */
new_phi = atan2(A_[m].imag, A_[m].real+1E-12);
model->phi[m] = new_phi;
}
}

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/*---------------------------------------------------------------------------*\
FILE........: phase.h
AUTHOR......: David Rowe
DATE CREATED: 1/2/09
Functions for modelling phase.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __PHASE__
#define __PHASE__
void phase_synth_zero_order(MODEL *model, float aks[], float *ex_phase);
#endif

131
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/*---------------------------------------------------------------------------*\
FILE........: postfilter.c
AUTHOR......: David Rowe
DATE CREATED: 13/09/09
Postfilter to improve sound quality for speech with high levels of
background noise. Unlike mixed-excitation models requires no bits
to be transmitted to handle background noise.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include "defines.h"
#include "dump.h"
#include "postfilter.h"
/*---------------------------------------------------------------------------*\
DEFINES
\*---------------------------------------------------------------------------*/
#define BG_THRESH 40.0 /* only consider low levels signals for bg_est */
#define BG_BETA 0.1 /* averaging filter constant */
/*---------------------------------------------------------------------------*\
postfilter()
The post filter is designed to help with speech corrupted by
background noise. The zero phase model tends to make speech with
background noise sound "clicky". With high levels of background
noise the low level inter-formant parts of the spectrum will contain
noise rather than speech harmonics, so modelling them as voiced
(i.e. a continuous, non-random phase track) is inaccurate.
Some codecs (like MBE) have a mixed voicing model that breaks the
spectrum into voiced and unvoiced regions. Several bits/frame
(5-12) are required to transmit the frequency selective voicing
information. Mixed excitation also requires accurate voicing
estimation (parameter estimators always break occasionally under
exceptional condition).
In our case we use a post filter approach which requires no
additional bits to be transmitted. The decoder measures the average
level of the background noise during unvoiced frames. If a harmonic
is less than this level it is made unvoiced by randomising it's
phases.
This idea is rather experimental. Some potential problems that may
happen:
1/ If someone says "aaaaaaaahhhhhhhhh" will background estimator track
up to speech level? This would be a bad thing.
2/ If background noise suddenly dissapears from the source speech does
estimate drop quickly? What is noise suddenly re-appears?
3/ Background noise with a non-flat sepctrum. Current algorithm just
comsiders scpetrum as a whole, but this could be broken up into
bands, each with their own estimator.
4/ Males and females with the same level of background noise. Check
performance the same. Changing Wo affects width of each band, may
affect bg energy estimates.
5/ Not sure what happens during long periods of voiced speech
e.g. "sshhhhhhh"
\*---------------------------------------------------------------------------*/
void postfilter(
MODEL *model,
float *bg_est
)
{
int m, uv;
float e;
/* determine average energy across spectrum */
e = 0.0;
for(m=1; m<=model->L; m++)
e += model->A[m]*model->A[m];
e = 10.0*log10(e/model->L);
/* If beneath threhold, update bg estimate. The idea
of the threshold is to prevent updating during high level
speech. */
if ((e < BG_THRESH) && !model->voiced)
*bg_est = *bg_est*(1.0 - BG_BETA) + e*BG_BETA;
/* now mess with phases during voiced frames to make any harmonics
less then our background estimate unvoiced.
*/
uv = 0;
if (model->voiced)
for(m=1; m<=model->L; m++)
if (20.0*log10(model->A[m]) < *bg_est) {
model->phi[m] = TWO_PI*(float)rand()/RAND_MAX;
uv++;
}
dump_bg(e, *bg_est, 100.0*uv/model->L);
}

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/*---------------------------------------------------------------------------*\
FILE........: postfilter.h
AUTHOR......: David Rowe
DATE CREATED: 13/09/09
Postfilter header file.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __POSTFILTER__
#define __POSTFILTER__
void postfilter(MODEL *model, float *bg_est);
#endif

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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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#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 "four1.h"
#include "codebook.h"
#define LSP_DELTA1 0.01 /* grid spacing for LSP root searches */
#define MAX_CB 20 /* max number of codebooks */
/* describes each codebook */
typedef struct {
int k; /* dimension of vector */
int log2m; /* number of bits in m */
int m; /* elements in codebook */
float *fn; /* file name of text file storing the VQ */
} LSP_CB;
/* lsp_q describes entire quantiser made up of several codebooks */
#ifdef OLDER
/* 10+10+6+6 = 32 bit LSP difference split VQ */
LSP_CB lsp_q[] = {
{3, 1024, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd123.txt"},
{3, 1024, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd456.txt"},
{2, 64, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd78.txt"},
{2, 64, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd910.txt"},
{0, 0, ""}
};
#endif
LSP_CB lsp_q[] = {
{1,4,16, codebook_lsp1 },
{1,4,16, codebook_lsp2 },
{1,4,16, codebook_lsp3 },
{1,4,16, codebook_lsp4 },
{1,4,16, codebook_lsp5 },
{1,4,16, codebook_lsp6 },
{1,4,16, codebook_lsp7 },
{1,3,8, codebook_lsp8 },
{1,3,8, codebook_lsp9 },
{1,2,4, codebook_lsp10 },
{0,0,0, NULL },
};
/* ptr to each codebook */
static float *plsp_cb[MAX_CB];
/*---------------------------------------------------------------------------*\
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_q[i].log2m;
}
/*---------------------------------------------------------------------------*\
quantise_uniform
Simulates uniform quantising of a float.
\*---------------------------------------------------------------------------*/
void quantise_uniform(float *val, float min, float max, int bits)
{
int levels = 1 << (bits-1);
float norm;
int index;
/* hard limit to quantiser range */
printf("min: %f max: %f val: %f ", min, max, val[0]);
if (val[0] < min) val[0] = min;
if (val[0] > max) val[0] = max;
norm = (*val - min)/(max-min);
printf("%f norm: %f ", val[0], norm);
index = fabs(levels*norm + 0.5);
*val = min + index*(max-min)/levels;
printf("index %d val_: %f\n", index, val[0]);
}
/*---------------------------------------------------------------------------*\
lspd_quantise
Simulates differential lsp quantiser
\*---------------------------------------------------------------------------*/
void lsp_quantise(
float lsp[],
float lsp_[],
int order
)
{
int i;
float dlsp[LPC_MAX];
float dlsp_[LPC_MAX];
dlsp[0] = lsp[0];
for(i=1; i<order; i++)
dlsp[i] = lsp[i] - lsp[i-1];
for(i=0; i<order; i++)
dlsp_[i] = dlsp[i];
quantise_uniform(&dlsp_[0], 0.1, 0.5, 5);
lsp_[0] = dlsp_[0];
for(i=1; i<order; i++)
lsp_[i] = lsp_[i-1] + dlsp_[i];
}
/*---------------------------------------------------------------------------*\
scan_line()
This function reads a vector of floats from a line in a text file.
\*---------------------------------------------------------------------------*/
void scan_line(FILE *fp, float f[], int n)
/* FILE *fp; file ptr to text file */
/* float f[]; array of floats to return */
/* int n; number of floats in line */
{
char s[MAX_STR];
char *ps,*pe;
int i;
fgets(s,MAX_STR,fp);
ps = pe = s;
for(i=0; i<n; i++) {
while( isspace(*pe)) pe++;
while( !isspace(*pe)) pe++;
sscanf(ps,"%f",&f[i]);
ps = pe;
}
}
/*---------------------------------------------------------------------------*\
load_cb
Loads a single codebook (LSP vector quantiser) into memory.
\*---------------------------------------------------------------------------*/
void load_cb(float *source, float *cb, int k, int m)
{
int lines;
int i;
lines = 0;
for(i=0; i<m; i++) {
cb[k*lines++] = source[i];
}
}
/*---------------------------------------------------------------------------*\
quantise_init
Loads the entire LSP quantiser comprised of several vector quantisers
(codebooks).
\*---------------------------------------------------------------------------*/
void quantise_init()
{
int i,k,m;
i = 0;
while(lsp_q[i].k) {
k = lsp_q[i].k;
m = lsp_q[i].m;
plsp_cb[i] = (float*)malloc(sizeof(float)*k*m);
assert(plsp_cb[i] != NULL);
load_cb(lsp_q[i].fn, plsp_cb[i], k, m);
i++;
assert(i < MAX_CB);
}
}
/*---------------------------------------------------------------------------*\
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(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;
besti = 0;
beste = 1E32;
for(j=0; j<m; j++) {
e = 0.0;
for(i=0; i<k; i++)
e += pow((cb[j*k+i]-vec[i])*w[i],2.0);
if (e < beste) {
beste = e;
besti = j;
}
}
*se += beste;
return(besti);
}
static float gmin=PI;
float get_gmin(void) { return gmin; }
void min_lsp_dist(float lsp[], int order)
{
int i;
for(i=1; i<order; i++)
if ((lsp[i]-lsp[i-1]) < gmin)
gmin = lsp[i]-lsp[i-1];
}
void check_lsp_order(float lsp[], int lpc_order)
{
int i;
float tmp;
for(i=1; i<lpc_order; i++)
if (lsp[i] < lsp[i-1]) {
printf("swap %d\n",i);
tmp = lsp[i-1];
lsp[i-1] = lsp[i]-0.05;
lsp[i] = tmp+0.05;
}
}
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;
}
}
/*---------------------------------------------------------------------------*\
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 */
int index;
float se;
int k,m;
float *cb;
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];
/* simple uniform scalar quantisers */
for(i=0; i<10; i++) {
k = lsp_q[i].k;
m = lsp_q[i].m;
cb = plsp_cb[i];
index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
lsp_hz[i] = cb[index*k];
}
/* 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 setp 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);
dump_lsp(lsp);
}
dump_E(E);
#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); /* {ak} -> {Am} LPC decode */
return snr;
}
/*---------------------------------------------------------------------------*\
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(
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 */
)
{
COMP Pw[FFT_DEC]; /* 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_DEC);
/* Determine DFT of A(exp(jw)) --------------------------------------------*/
for(i=0; i<FFT_DEC; i++) {
Pw[i].real = 0.0;
Pw[i].imag = 0.0;
}
for(i=0; i<=order; i++)
Pw[i].real = ak[i];
four1(&Pw[-1].imag,FFT_DEC,1);
/* Determine power spectrum P(w) = E/(A(exp(jw))^2 ------------------------*/
for(i=0; i<FFT_DEC/2; i++)
Pw[i].real = E/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
if (dump)
dump_Pw(Pw);
/* Determine magnitudes by linear interpolation of P(w) -------------------*/
signal = noise = 0.0;
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);
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....: 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;
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];
roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
// assert(roots == order);
return E;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_lsps()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
From a vector of unquantised (floating point) LSPs finds the quantised
LSP indexes.
\*---------------------------------------------------------------------------*/
void encode_lsps(int indexes[], float lsp[], int order)
{
int i,k,m;
float wt[1];
float lsp_hz[LPC_MAX];
float *cb, 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];
/* simple uniform scalar quantisers */
wt[0] = 1.0;
for(i=0; i<order; i++) {
k = lsp_q[i].k;
m = lsp_q[i].m;
cb = plsp_cb[i];
indexes[i] = quantise(cb, &lsp_hz[i], wt, k, m, &se);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_lsps()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
From a vector of quantised LSP indexes, returns the quantised
(floating point) LSPs.
\*---------------------------------------------------------------------------*/
void decode_lsps(float lsp[], int indexes[], int order)
{
int i,k;
float lsp_hz[LPC_MAX];
float *cb;
for(i=0; i<order; i++) {
k = lsp_q[i].k;
cb = plsp_cb[i];
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....: 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 setp
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<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. This may need more experiment for
different quanstisers.
*/
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);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: need_lpc_correction()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Determine if we need LPC correction of first harmonic.
\*---------------------------------------------------------------------------*/
int need_lpc_correction(MODEL *model, float ak[], float E)
{
MODEL tmp;
float snr,E1;
/* Find amplitudes so we can check if we need LPC correction.
TODO: replace call to aks_to_M2() by a single DFT calculation
of E/A(exp(jWo)) to make much more efficient. We only need
A[1].
*/
memcpy(&tmp, model, sizeof(MODEL));
aks_to_M2(ak, LPC_ORD, &tmp, E, &snr, 0);
/*
Attenuate fundamental by 30dB if F0 < 150 Hz and LPC modelling
error for A[1] is larger than 6dB.
LPC modelling often makes big errors on 1st harmonic, for example
when the fundamental has been removed by analog high pass
filtering before sampling. However on unfiltered speech from
high quality sources we would like to keep the fundamental to
maintain the speech quality. So we check the error in A[1] and
attenuate it if the error is large to avoid annoying low
frequency energy after LPC modelling.
This requires a single bit to quantise, on top of the other
spectral magnitude bits (i.e. LSP bits + 1 total).
*/
E1 = fabs(20.0*log10(model->A[1]) - 20.0*log10(tmp.A[1]));
if (E1 > 6.0)
return 1;
else
return 0;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: apply_lpc_correction()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Apply first harmonic LPC correction at decoder.
\*---------------------------------------------------------------------------*/
void apply_lpc_correction(MODEL *model, int lpc_correction)
{
if (lpc_correction) {
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 WO_BITS 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;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: encode_amplitudes()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Time domain LPC is used model the amplitudes which are then
converted to LSPs and quantised. So we don't actually encode the
amplitudes directly, rather we derive an equivalent representation
from the time domain speech.
\*---------------------------------------------------------------------------*/
void encode_amplitudes(int lsp_indexes[],
int *lpc_correction,
int *energy_index,
MODEL *model,
float Sn[],
float w[])
{
float lsps[LPC_ORD];
float ak[LPC_ORD+1];
float e;
e = speech_to_uq_lsps(lsps, ak, Sn, w, LPC_ORD);
encode_lsps(lsp_indexes, lsps, LPC_ORD);
*lpc_correction = need_lpc_correction(model, ak, e);
*energy_index = encode_energy(e);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: decode_amplitudes()
AUTHOR......: David Rowe
DATE CREATED: 22/8/2010
Given the amplitude quantiser indexes recovers the harmonic
amplitudes.
\*---------------------------------------------------------------------------*/
float decode_amplitudes(MODEL *model,
float ak[],
int lsp_indexes[],
int lpc_correction,
int energy_index
)
{
float lsps[LPC_ORD];
float e;
float snr;
decode_lsps(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);
apply_lpc_correction(model, lpc_correction);
return snr;
}

84
libs/libcodec2/src/quantise.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: quantise.h
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __QUANTISE__
#define __QUANTISE__
#define WO_BITS 7
#define WO_LEVELS (1<<WO_BITS)
#define E_BITS 5
#define E_LEVELS (1<<E_BITS)
#define E_MIN_DB -10.0
#define E_MAX_DB 40.0
void quantise_init();
float lpc_model_amplitudes(float Sn[], float w[], MODEL *model, int order,
int lsp,float ak[]);
void aks_to_M2(float ak[], int order, MODEL *model, float E, float *snr,
int dump);
float get_gmin(void);
int encode_Wo(float Wo);
float decode_Wo(int index);
void encode_lsps(int indexes[], float lsp[], int order);
void decode_lsps(float lsp[], int indexes[], int order);
int encode_energy(float e);
float decode_energy(int index);
void encode_amplitudes(int lsp_indexes[],
int *lpc_correction,
int *energy_index,
MODEL *model,
float Sn[],
float w[]);
float decode_amplitudes(MODEL *model,
float ak[],
int lsp_indexes[],
int lpc_correction,
int energy_index);
void pack(unsigned char * bits, unsigned int *nbit, int index, unsigned int index_bits);
int unpack(const unsigned char * bits, unsigned int *nbit, unsigned int index_bits);
int lsp_bits(int i);
int need_lpc_correction(MODEL *model, float ak[], float E);
void apply_lpc_correction(MODEL *model, int lpc_correction);
float speech_to_uq_lsps(float lsp[],
float ak[],
float Sn[],
float w[],
int order
);
void bw_expand_lsps(float lsp[],
int order
);
void decode_lsps(float lsp[], int indexes[], int order);
#endif

18
libs/libcodec2/src/sim.sh Executable file
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#!/bin/sh
# sim.sh
# David Rowe 10 Sep 2009
# Process a source file using the codec 2 simulation. An output
# speech file is generated for each major processing step, from the
# unquantised siusoidal model to fully quantised. This way we can
# listen to the effect of each processing step. Use listensim.sh to
# test the output files.
../src/c2sim ../raw/$1.raw -o $1_uq.raw
../src/c2sim ../raw/$1.raw --phase0 -o $1_phase0.raw --postfilter
../src/c2sim ../raw/$1.raw --lpc 10 -o $1_lpc10.raw
../src/c2sim ../raw/$1.raw --lpc 10 --lsp -o $1_lsp.raw
../src/c2sim ../raw/$1.raw --phase0 --lpc 10 -o $1_phase0_lpc10.raw --postfilter
../src/c2sim ../raw/$1.raw --phase0 --lpc 10 --lsp -o $1_phase0_lsp.raw --postfilter
../src/c2sim ../raw/$1.raw --phase0 --lpc 10 --lsp -o $1_phase0_lsp_dec.raw --postfilter --dec

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libs/libcodec2/src/sine.c Normal file
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/*---------------------------------------------------------------------------*\
FILE........: sine.c
AUTHOR......: David Rowe
DATE CREATED: 19/8/2010
Sinusoidal analysis and synthesis functions.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 1990-2010 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
/*---------------------------------------------------------------------------*\
INCLUDES
\*---------------------------------------------------------------------------*/
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include "defines.h"
#include "sine.h"
#include "four1.h"
/*---------------------------------------------------------------------------*\
HEADERS
\*---------------------------------------------------------------------------*/
void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
float pstep);
/*---------------------------------------------------------------------------*\
FUNCTIONS
\*---------------------------------------------------------------------------*/
/*---------------------------------------------------------------------------*\
FUNCTION....: make_analysis_window
AUTHOR......: David Rowe
DATE CREATED: 11/5/94
Init function that generates the time domain analysis window and it's DFT.
\*---------------------------------------------------------------------------*/
void make_analysis_window(float w[],COMP W[])
{
float m;
COMP temp;
int i,j;
/*
Generate Hamming window centered on M-sample pitch analysis window
0 M/2 M-1
|-------------|-------------|
|-------|-------|
NW samples
All our analysis/synthsis is centred on the M/2 sample.
*/
m = 0.0;
for(i=0; i<M/2-NW/2; i++)
w[i] = 0.0;
for(i=M/2-NW/2,j=0; i<M/2+NW/2; i++,j++) {
w[i] = 0.5 - 0.5*cos(TWO_PI*j/(NW-1));
m += w[i]*w[i];
}
for(i=M/2+NW/2; i<M; i++)
w[i] = 0.0;
/* Normalise - makes freq domain amplitude estimation straight
forward */
m = 1.0/sqrt(m*FFT_ENC);
for(i=0; i<M; i++) {
w[i] *= m;
}
/*
Generate DFT of analysis window, used for later processing. Note
we modulo FFT_ENC shift the time domain window w[], this makes the
imaginary part of the DFT W[] equal to zero as the shifted w[] is
even about the n=0 time axis if NW is odd. Having the imag part
of the DFT W[] makes computation easier.
0 FFT_ENC-1
|-------------------------|
----\ /----
\ /
\ / <- shifted version of window w[n]
\ /
\ /
-------
|---------| |---------|
NW/2 NW/2
*/
for(i=0; i<FFT_ENC; i++) {
W[i].real = 0.0;
W[i].imag = 0.0;
}
for(i=0; i<NW/2; i++)
W[i].real = w[i+M/2];
for(i=FFT_ENC-NW/2,j=M/2-NW/2; i<FFT_ENC; i++,j++)
W[i].real = w[j];
four1(&W[-1].imag,FFT_ENC,-1); /* "Numerical Recipes in C" FFT */
/*
Re-arrange W[] to be symmetrical about FFT_ENC/2. Makes later
analysis convenient.
Before:
0 FFT_ENC-1
|----------|---------|
__ _
\ /
\_______________/
After:
0 FFT_ENC-1
|----------|---------|
___
/ \
________/ \_______
*/
for(i=0; i<FFT_ENC/2; i++) {
temp.real = W[i].real;
temp.imag = W[i].imag;
W[i].real = W[i+FFT_ENC/2].real;
W[i].imag = W[i+FFT_ENC/2].imag;
W[i+FFT_ENC/2].real = temp.real;
W[i+FFT_ENC/2].imag = temp.imag;
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: dft_speech
AUTHOR......: David Rowe
DATE CREATED: 27/5/94
Finds the DFT of the current speech input speech frame.
\*---------------------------------------------------------------------------*/
void dft_speech(COMP Sw[], float Sn[], float w[])
{
int i;
for(i=0; i<FFT_ENC; i++) {
Sw[i].real = 0.0;
Sw[i].imag = 0.0;
}
/* Centre analysis window on time axis, we need to arrange input
to FFT this way to make FFT phases correct */
/* move 2nd half to start of FFT input vector */
for(i=0; i<NW/2; i++)
Sw[i].real = Sn[i+M/2]*w[i+M/2];
/* move 1st half to end of FFT input vector */
for(i=0; i<NW/2; i++)
Sw[FFT_ENC-NW/2+i].real = Sn[i+M/2-NW/2]*w[i+M/2-NW/2];
four1(&Sw[-1].imag,FFT_ENC,-1);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: two_stage_pitch_refinement
AUTHOR......: David Rowe
DATE CREATED: 27/5/94
Refines the current pitch estimate using the harmonic sum pitch
estimation technique.
\*---------------------------------------------------------------------------*/
void two_stage_pitch_refinement(MODEL *model, COMP Sw[])
{
float pmin,pmax,pstep; /* pitch refinment minimum, maximum and step */
/* Coarse refinement */
pmax = TWO_PI/model->Wo + 5;
pmin = TWO_PI/model->Wo - 5;
pstep = 1.0;
hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
/* Fine refinement */
pmax = TWO_PI/model->Wo + 1;
pmin = TWO_PI/model->Wo - 1;
pstep = 0.25;
hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
/* Limit range */
if (model->Wo < TWO_PI/P_MAX)
model->Wo = TWO_PI/P_MAX;
if (model->Wo > TWO_PI/P_MIN)
model->Wo = TWO_PI/P_MIN;
model->L = floor(PI/model->Wo);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: hs_pitch_refinement
AUTHOR......: David Rowe
DATE CREATED: 27/5/94
Harmonic sum pitch refinement function.
pmin pitch search range minimum
pmax pitch search range maximum
step pitch search step size
model current pitch estimate in model.Wo
model refined pitch estimate in model.Wo
\*---------------------------------------------------------------------------*/
void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float pstep)
{
int m; /* loop variable */
int b; /* bin for current harmonic centre */
float E; /* energy for current pitch*/
float Wo; /* current "test" fundamental freq. */
float Wom; /* Wo that maximises E */
float Em; /* mamimum energy */
float r; /* number of rads/bin */
float p; /* current pitch */
/* Initialisation */
model->L = PI/model->Wo; /* use initial pitch est. for L */
Wom = model->Wo;
Em = 0.0;
r = TWO_PI/FFT_ENC;
/* Determine harmonic sum for a range of Wo values */
for(p=pmin; p<=pmax; p+=pstep) {
E = 0.0;
Wo = TWO_PI/p;
/* Sum harmonic magnitudes */
for(m=1; m<=model->L; m++) {
b = floor(m*Wo/r + 0.5);
E += Sw[b].real*Sw[b].real + Sw[b].imag*Sw[b].imag;
}
/* Compare to see if this is a maximum */
if (E > Em) {
Em = E;
Wom = Wo;
}
}
model->Wo = Wom;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: estimate_amplitudes
AUTHOR......: David Rowe
DATE CREATED: 27/5/94
Estimates the complex amplitudes of the harmonics.
\*---------------------------------------------------------------------------*/
void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[])
{
int i,m; /* loop variables */
int am,bm; /* bounds of current harmonic */
int b; /* DFT bin of centre of current harmonic */
float den; /* denominator of amplitude expression */
float r; /* number of rads/bin */
int offset;
COMP Am;
r = TWO_PI/FFT_ENC;
for(m=1; m<=model->L; m++) {
den = 0.0;
am = floor((m - 0.5)*model->Wo/r + 0.5);
bm = floor((m + 0.5)*model->Wo/r + 0.5);
b = floor(m*model->Wo/r + 0.5);
/* Estimate ampltude of harmonic */
den = 0.0;
Am.real = Am.imag = 0.0;
for(i=am; i<bm; i++) {
den += Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag;
offset = i + FFT_ENC/2 - floor(m*model->Wo/r + 0.5);
Am.real += Sw[i].real*W[offset].real;
Am.imag += Sw[i].imag*W[offset].real;
}
model->A[m] = sqrt(den);
/* Estimate phase of harmonic */
model->phi[m] = atan2(Sw[b].imag,Sw[b].real);
}
}
/*---------------------------------------------------------------------------*\
est_voicing_mbe()
Returns the error of the MBE cost function for a fiven F0.
Note: I think a lot of the operations below can be simplified as
W[].imag = 0 and has been normalised such that den always equals 1.
\*---------------------------------------------------------------------------*/
float est_voicing_mbe(
MODEL *model,
COMP Sw[],
COMP W[],
float f0,
COMP Sw_[] /* DFT of all voiced synthesised signal for f0 */
/* useful for debugging/dump file */
)
{
int i,l,al,bl,m; /* loop variables */
COMP Am; /* amplitude sample for this band */
int offset; /* centers Hw[] about current harmonic */
float den; /* denominator of Am expression */
float error; /* accumulated error between originl and synthesised */
float Wo; /* current "test" fundamental freq. */
int L;
float sig, snr;
sig = 0.0;
for(l=1; l<=model->L/4; l++) {
sig += model->A[l]*model->A[l];
}
for(i=0; i<FFT_ENC; i++) {
Sw_[i].real = 0.0;
Sw_[i].imag = 0.0;
}
L = floor((FS/2.0)/f0);
Wo = f0*(TWO_PI/FS);
error = 0.0;
/* Just test across the harmonics in the first 1000 Hz (L/4) */
for(l=1; l<=L/4; l++) {
Am.real = 0.0;
Am.imag = 0.0;
den = 0.0;
al = ceil((l - 0.5)*Wo*FFT_ENC/TWO_PI);
bl = ceil((l + 0.5)*Wo*FFT_ENC/TWO_PI);
/* Estimate amplitude of harmonic assuming harmonic is totally voiced */
for(m=al; m<bl; m++) {
offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
Am.real += Sw[m].real*W[offset].real + Sw[m].imag*W[offset].imag;
Am.imag += Sw[m].imag*W[offset].real - Sw[m].real*W[offset].imag;
den += W[offset].real*W[offset].real + W[offset].imag*W[offset].imag;
}
Am.real = Am.real/den;
Am.imag = Am.imag/den;
/* Determine error between estimated harmonic and original */
for(m=al; m<bl; m++) {
offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
Sw_[m].real = Am.real*W[offset].real - Am.imag*W[offset].imag;
Sw_[m].imag = Am.real*W[offset].imag + Am.imag*W[offset].real;
error += (Sw[m].real - Sw_[m].real)*(Sw[m].real - Sw_[m].real);
error += (Sw[m].imag - Sw_[m].imag)*(Sw[m].imag - Sw_[m].imag);
}
}
snr = 10.0*log10(sig/error);
if (snr > V_THRESH)
model->voiced = 1;
else
model->voiced = 0;
return snr;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: make_synthesis_window
AUTHOR......: David Rowe
DATE CREATED: 11/5/94
Init function that generates the trapezoidal (Parzen) sythesis window.
\*---------------------------------------------------------------------------*/
void make_synthesis_window(float Pn[])
{
int i;
float win;
/* Generate Parzen window in time domain */
win = 0.0;
for(i=0; i<N/2-TW; i++)
Pn[i] = 0.0;
win = 0.0;
for(i=N/2-TW; i<N/2+TW; win+=1.0/(2*TW), i++ )
Pn[i] = win;
for(i=N/2+TW; i<3*N/2-TW; i++)
Pn[i] = 1.0;
win = 1.0;
for(i=3*N/2-TW; i<3*N/2+TW; win-=1.0/(2*TW), i++)
Pn[i] = win;
for(i=3*N/2+TW; i<2*N; i++)
Pn[i] = 0.0;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: synthesise
AUTHOR......: David Rowe
DATE CREATED: 20/2/95
Synthesise a speech signal in the frequency domain from the
sinusodal model parameters. Uses overlap-add a triangular window to
smoothly interpolate betwen frames.
\*---------------------------------------------------------------------------*/
void synthesise(
float Sn_[], /* time domain synthesised signal */
MODEL *model, /* ptr to model parameters for this frame */
float Pn[], /* time domain Parzen window */
int shift /* used to handle transition frames */
)
{
int i,l,j,b; /* loop variables */
COMP Sw_[FFT_DEC]; /* DFT of synthesised signal */
if (shift) {
/* Update memories */
for(i=0; i<N-1; i++) {
Sn_[i] = Sn_[i+N];
}
Sn_[N-1] = 0.0;
}
for(i=0; i<FFT_DEC; i++) {
Sw_[i].real = 0.0;
Sw_[i].imag = 0.0;
}
/* Now set up frequency domain synthesised speech */
for(l=1; l<=model->L; l++) {
b = floor(l*model->Wo*FFT_DEC/TWO_PI + 0.5);
Sw_[b].real = model->A[l]*cos(model->phi[l]);
Sw_[b].imag = model->A[l]*sin(model->phi[l]);
Sw_[FFT_DEC-b].real = Sw_[b].real;
Sw_[FFT_DEC-b].imag = -Sw_[b].imag;
}
/* Perform inverse DFT */
four1(&Sw_[-1].imag,FFT_DEC,1);
/* Overlap add to previous samples */
for(i=0; i<N-1; i++) {
Sn_[i] += Sw_[FFT_DEC-N+1+i].real*Pn[i];
}
if (shift)
for(i=N-1,j=0; i<2*N; i++,j++)
Sn_[i] = Sw_[j].real*Pn[i];
else
for(i=N-1,j=0; i<2*N; i++,j++)
Sn_[i] += Sw_[j].real*Pn[i];
}

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libs/libcodec2/src/sine.h Normal file
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/*---------------------------------------------------------------------------*\
FILE........: sine.h
AUTHOR......: David Rowe
DATE CREATED: 1/11/94
Header file for sinusoidal analysis and synthesis functions.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
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, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __SINE__
#define __SINE__
void make_analysis_window(float w[], COMP W[]);
void dft_speech(COMP Sw[], float Sn[], float w[]);
void two_stage_pitch_refinement(MODEL *model, COMP Sw[]);
void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[]);
float est_voicing_mbe(MODEL *model, COMP Sw[], COMP W[], float f0, COMP Sw_[]);
void make_synthesis_window(float Pn[]);
void synthesise(float Sn_[], MODEL *model, float Pn[], int shift);
#endif