rm-ed 450 & 450WB

pull/3/head
drowe67 2023-07-16 10:14:55 +09:30 committed by David Rowe
parent 9f5e2deebc
commit 6549fa1d6f
7 changed files with 9 additions and 852 deletions

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@ -1146,14 +1146,6 @@ endif(NOT APPLE)
COMMAND sh -c "./c2enc 700C ${CMAKE_CURRENT_SOURCE_DIR}/raw/hts1a.raw - | ./c2dec 700C - - | sox -t .s16 -r 8000 - hts1a_700C.wav"
WORKING_DIRECTORY ${CMAKE_CURRENT_BINARY_DIR}/src
)
add_test(NAME test_codec2_mode_450
COMMAND sh -c "./c2enc 450 ${CMAKE_CURRENT_SOURCE_DIR}/raw/hts1a.raw - | ./c2dec 450 - - | sox -t .s16 -r 8000 - hts1a_450.wav"
WORKING_DIRECTORY ${CMAKE_CURRENT_BINARY_DIR}/src
)
add_test(NAME test_codec2_mode_450PWB
COMMAND sh -c "./c2enc 450PWB ${CMAKE_CURRENT_SOURCE_DIR}/raw/hts1a.raw - | ./c2dec 450PWB - - | sox -t .s16 -r 16000 - hts1a_450PWB.wav"
WORKING_DIRECTORY ${CMAKE_CURRENT_BINARY_DIR}/src
)
add_test(NAME test_vq_mbest
COMMAND sh -c "./tvq_mbest; \

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@ -169,7 +169,6 @@ set(CODEC2_SRCS
lsp.c
mbest.c
newamp1.c
newamp2.c
ofdm.c
ofdm_mode.c
phase.c

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@ -139,14 +139,10 @@ int main(int argc, char *argv[]) {
mode = CODEC2_MODE_1200;
else if (strcmp(argv[1], "700C") == 0)
mode = CODEC2_MODE_700C;
else if (strcmp(argv[1], "450") == 0)
mode = CODEC2_MODE_450;
else if (strcmp(argv[1], "450PWB") == 0)
mode = CODEC2_MODE_450PWB;
else {
fprintf(stderr,
"Error in mode: %s. Must be 3200, 2400, 1600, 1400, 1300, 1200, "
"700C, 450, or 450PWB\n",
"700C\n",
argv[1]);
exit(1);
}
@ -396,7 +392,7 @@ void print_help(const struct option *long_options, int num_opts, char *argv[]) {
char *option_parameters;
fprintf(stderr,
"\nc2dec - Codec 2 decoder and bit error simulation program\n"
"usage: %s 3200|2400|1600|1400|1300|1200|700C|450|450PWB InputFile "
"usage: %s 3200|2400|1600|1400|1300|1200|700C InputFile "
"OutputRawFile [OPTIONS]\n\n"
"Options:\n",
argv[0]);

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@ -51,7 +51,7 @@ int main(int argc, char *argv[]) {
if (argc < 4) {
printf(
"usage: c2enc 3200|2400|1600|1400|1300|1200|700C|450|450PWB "
"usage: c2enc 3200|2400|1600|1400|1300|1200|700C "
"InputRawspeechFile OutputBitFile [--natural] [--softdec] "
"[--bitperchar] [--mlfeat f32File modelFile] [--loadcb stageNum "
"Filename] [--var] [--eq]\n");
@ -76,14 +76,10 @@ int main(int argc, char *argv[]) {
mode = CODEC2_MODE_1200;
else if (strcmp(argv[1], "700C") == 0)
mode = CODEC2_MODE_700C;
else if (strcmp(argv[1], "450") == 0)
mode = CODEC2_MODE_450;
else if (strcmp(argv[1], "450PWB") == 0)
mode = CODEC2_MODE_450;
else {
fprintf(stderr,
"Error in mode: %s. Must be 3200, 2400, 1600, 1400, 1300, 1200, "
"700C, 450, 450PWB or WB\n",
"700C\n",
argv[1]);
exit(1);
}

View File

@ -83,11 +83,6 @@ void codec2_decode_1200(struct CODEC2 *c2, short speech[],
void codec2_encode_700c(struct CODEC2 *c2, unsigned char *bits, short speech[]);
void codec2_decode_700c(struct CODEC2 *c2, short speech[],
const unsigned char *bits);
void codec2_encode_450(struct CODEC2 *c2, unsigned char *bits, short speech[]);
void codec2_decode_450(struct CODEC2 *c2, short speech[],
const unsigned char *bits);
void codec2_decode_450pwb(struct CODEC2 *c2, short speech[],
const unsigned char *bits);
static void ear_protection(float in_out[], int n);
/*---------------------------------------------------------------------------*\
@ -110,7 +105,6 @@ static void ear_protection(float in_out[], int n);
\*---------------------------------------------------------------------------*/
// Don't create CODEC2_MODE_450PWB for Encoding as it has undefined behavior !
struct CODEC2 *codec2_create(int mode) {
struct CODEC2 *c2;
int i, l;
@ -125,9 +119,7 @@ struct CODEC2 *codec2_create(int mode) {
CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_450, mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode))) {
CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, mode))) {
return NULL;
}
@ -138,11 +130,7 @@ struct CODEC2 *codec2_create(int mode) {
/* store constants in a few places for convenience */
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode) == 0) {
c2->c2const = c2const_create(8000, N_S);
} else {
c2->c2const = c2const_create(16000, N_S);
}
c2->c2const = c2const_create(8000, N_S);
c2->Fs = c2->c2const.Fs;
int n_samp = c2->n_samp = c2->c2const.n_samp;
int m_pitch = c2->m_pitch = c2->c2const.m_pitch;
@ -231,40 +219,10 @@ struct CODEC2 *codec2_create(int mode) {
c2->eq_en = false;
c2->Wo_left = 0.0;
c2->voicing_left = 0;
;
c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 0, NULL, NULL);
c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 1, NULL, NULL);
}
/* newamp2 initialisation */
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) {
n2_mel_sample_freqs_kHz(c2->n2_rate_K_sample_freqs_kHz, NEWAMP2_K);
int k;
for (k = 0; k < NEWAMP2_K; k++) {
c2->n2_prev_rate_K_vec_[k] = 0.0;
}
c2->Wo_left = 0.0;
c2->voicing_left = 0;
;
c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 0, NULL, NULL);
c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 1, NULL, NULL);
}
/* newamp2 PWB initialisation */
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) {
n2_mel_sample_freqs_kHz(c2->n2_pwb_rate_K_sample_freqs_kHz, NEWAMP2_16K_K);
int k;
for (k = 0; k < NEWAMP2_16K_K; k++) {
c2->n2_pwb_prev_rate_K_vec_[k] = 0.0;
}
c2->Wo_left = 0.0;
c2->voicing_left = 0;
;
c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 0, NULL, NULL);
c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 1, NULL, NULL);
}
c2->fmlfeat = NULL;
c2->fmlmodel = NULL;
@ -310,17 +268,6 @@ struct CODEC2 *codec2_create(int mode) {
c2->decode = codec2_decode_700c;
}
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) {
c2->encode = codec2_encode_450;
c2->decode = codec2_decode_450;
}
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) {
// Encode PWB doesn't make sense
c2->encode = codec2_encode_450;
c2->decode = codec2_decode_450pwb;
}
return c2;
}
@ -345,14 +292,6 @@ void codec2_destroy(struct CODEC2 *c2) {
codec2_fft_free(c2->phase_fft_fwd_cfg);
codec2_fft_free(c2->phase_fft_inv_cfg);
}
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) {
codec2_fft_free(c2->phase_fft_fwd_cfg);
codec2_fft_free(c2->phase_fft_inv_cfg);
}
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) {
codec2_fft_free(c2->phase_fft_fwd_cfg);
codec2_fft_free(c2->phase_fft_inv_cfg);
}
FREE(c2->Pn);
FREE(c2->Sn);
FREE(c2->w);
@ -378,8 +317,6 @@ int codec2_bits_per_frame(struct CODEC2 *c2) {
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) return 52;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) return 48;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) return 28;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) return 18;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) return 18;
return 0; /* shouldn't get here */
}
@ -417,8 +354,6 @@ int codec2_samples_per_frame(struct CODEC2 *c2) {
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) return 320;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) return 320;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) return 320;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) return 320;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) return 640;
return 0; /* shouldn't get here */
}
@ -1640,26 +1575,6 @@ float codec2_energy_700c(struct CODEC2 *c2, const unsigned char *bits) {
return POW10F(mean / 10.0);
}
float codec2_energy_450(struct CODEC2 *c2, const unsigned char *bits) {
int indexes[4];
unsigned int nbit = 0;
assert(c2 != NULL);
/* unpack bits from channel ------------------------------------*/
indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0);
// indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0);
indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0);
indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0);
float mean = newamp2_energy_cb[0].cb[indexes[2]];
mean -= 10;
if (indexes[3] == 0) mean -= 10;
return POW10F(mean / 10.0);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_get_energy()
@ -1678,9 +1593,7 @@ float codec2_get_energy(struct CODEC2 *c2, const unsigned char *bits) {
(CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) ||
(CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) ||
(CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) ||
(CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) ||
(CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) ||
(CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)));
(CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)));
MODEL model;
float xq_dec[2] = {};
int e_index, WoE_index;
@ -1720,203 +1633,10 @@ float codec2_get_energy(struct CODEC2 *c2, const unsigned char *bits) {
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) {
e = codec2_energy_700c(c2, bits);
}
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) {
e = codec2_energy_450(c2, bits);
}
return e;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_encode_450
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
450 bit/s codec that uses newamp2 fixed rate VQ of amplitudes.
Encodes 320 speech samples (40ms of speech) into 28 bits.
The codec2 algorithm actually operates internally on 10ms (80
sample) frames, so we run the encoding algorithm four times:
frame 0: nothing
frame 1: nothing
frame 2: nothing
frame 3: 9 bit 1 stage VQ, 3 bits energy,
6 bit scalar Wo/voicing/plosive. No spare bits.
If a plosive is detected the frame at the energy-step is encoded.
Voicing is encoded using the 000000 index of the Wo quantiser.
Plosive is encoded using the 111111 index of the Wo quantiser.
The bit allocation is:
Parameter frames 1-3 frame 4 Total
-----------------------------------------------------------
Harmonic magnitudes (rate k VQ) 0 9 9
Energy 0 3 3
log Wo/voicing/plosive 0 6 6
TOTAL 0 18 18
\*---------------------------------------------------------------------------*/
void codec2_encode_450(struct CODEC2 *c2, unsigned char *bits, short speech[]) {
MODEL model;
int indexes[4], i, h, M = 4;
unsigned int nbit = 0;
int plosiv = 0;
float energydelta[M];
int spectralCounter;
assert(c2 != NULL);
memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8));
for (i = 0; i < M; i++) {
analyse_one_frame(c2, &model, &speech[i * c2->n_samp]);
energydelta[i] = 0;
spectralCounter = 0;
for (h = 0; h < (model.L); h++) {
// only detect above 300 Hz
if (h * model.Wo * (c2->c2const.Fs / 2000.0) / M_PI > 0.3) {
energydelta[i] =
(double)energydelta[i] + (double)20.0 * log10(model.A[10] + 1E-16);
spectralCounter = spectralCounter + 1;
}
}
energydelta[i] = energydelta[i] / spectralCounter;
}
// Constants for plosive Detection tdB = threshold; minPwr = from below this
// level plosives have to rise
float tdB = 15; // not fixed can be changed
float minPwr = 15; // not fixed can be changed
if ((c2->energy_prev) < minPwr &&
energydelta[0] > ((c2->energy_prev) + tdB)) {
plosiv = 1;
}
if (energydelta[0] < minPwr && energydelta[1] > (energydelta[0] + tdB)) {
plosiv = 2;
}
if (energydelta[1] < minPwr && energydelta[2] > (energydelta[1] + tdB)) {
plosiv = 3;
}
if (energydelta[2] < minPwr && energydelta[3] > (energydelta[2] + tdB)) {
plosiv = 4;
}
if (plosiv != 0 && plosiv != 4) {
analyse_one_frame(c2, &model, &speech[(plosiv - 1) * c2->n_samp]);
}
c2->energy_prev = energydelta[3];
int K = 29;
float rate_K_vec[K], mean;
float rate_K_vec_no_mean[K], rate_K_vec_no_mean_[K];
if (plosiv > 0) {
plosiv = 1;
}
newamp2_model_to_indexes(&c2->c2const, indexes, &model, rate_K_vec,
c2->n2_rate_K_sample_freqs_kHz, K, &mean,
rate_K_vec_no_mean, rate_K_vec_no_mean_, plosiv);
pack_natural_or_gray(bits, &nbit, indexes[0], 9, 0);
// pack_natural_or_gray(bits, &nbit, indexes[1], 9, 0);
pack_natural_or_gray(bits, &nbit, indexes[2], 3, 0);
pack_natural_or_gray(bits, &nbit, indexes[3], 6, 0);
assert(nbit == (unsigned)codec2_bits_per_frame(c2));
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_decode_450
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
\*---------------------------------------------------------------------------*/
void codec2_decode_450(struct CODEC2 *c2, short speech[],
const unsigned char *bits) {
MODEL model[4];
int indexes[4];
int i;
unsigned int nbit = 0;
assert(c2 != NULL);
/* unpack bits from channel ------------------------------------*/
indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0);
// indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0);
indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0);
indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0);
int M = 4;
COMP HH[M][MAX_AMP + 1];
float interpolated_surface_[M][NEWAMP2_K];
int pwbFlag = 0;
newamp2_indexes_to_model(
&c2->c2const, model, (COMP *)HH, (float *)interpolated_surface_,
c2->n2_prev_rate_K_vec_, &c2->Wo_left, &c2->voicing_left,
c2->n2_rate_K_sample_freqs_kHz, NEWAMP2_K, c2->phase_fft_fwd_cfg,
c2->phase_fft_inv_cfg, indexes, 1.5, pwbFlag);
for (i = 0; i < M; i++) {
synthesise_one_frame(c2, &speech[c2->n_samp * i], &model[i], &HH[i][0],
1.5);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: codec2_decode_450pwb
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
Decodes the 450 codec data in pseudo wideband at 16kHz samplerate.
\*---------------------------------------------------------------------------*/
void codec2_decode_450pwb(struct CODEC2 *c2, short speech[],
const unsigned char *bits) {
MODEL model[4];
int indexes[4];
int i;
unsigned int nbit = 0;
assert(c2 != NULL);
/* unpack bits from channel ------------------------------------*/
indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0);
// indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0);
indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0);
indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0);
int M = 4;
COMP HH[M][MAX_AMP + 1];
float interpolated_surface_[M][NEWAMP2_16K_K];
int pwbFlag = 1;
newamp2_indexes_to_model(
&c2->c2const, model, (COMP *)HH, (float *)interpolated_surface_,
c2->n2_pwb_prev_rate_K_vec_, &c2->Wo_left, &c2->voicing_left,
c2->n2_pwb_rate_K_sample_freqs_kHz, NEWAMP2_16K_K, c2->phase_fft_fwd_cfg,
c2->phase_fft_inv_cfg, indexes, 1.5, pwbFlag);
for (i = 0; i < M; i++) {
synthesise_one_frame(c2, &speech[c2->n_samp * i], &model[i], &HH[i][0],
1.5);
}
}
/*---------------------------------------------------------------------------* \
FUNCTION....: synthesise_one_frame()
@ -1931,10 +1651,8 @@ void synthesise_one_frame(struct CODEC2 *c2, short speech[], MODEL *model,
COMP Aw[], float gain) {
int i;
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode) ||
CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) {
/* newamp1/2, we've already worked out rate L phase */
if (CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) {
/* newamp1, we've already worked out rate L phase */
COMP *H = Aw;
phase_synth_zero_order(c2->n_samp, model, &c2->ex_phase, H);
} else {

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@ -42,8 +42,6 @@ extern "C" {
#define CODEC2_MODE_1300 4
#define CODEC2_MODE_1200 5
#define CODEC2_MODE_700C 8
#define CODEC2_MODE_450 10
#define CODEC2_MODE_450PWB 11
#ifndef CODEC2_MODE_EN_DEFAULT
#define CODEC2_MODE_EN_DEFAULT 1
@ -78,12 +76,6 @@ extern "C" {
#if !defined(CODEC2_MODE_700C_EN)
#define CODEC2_MODE_700C_EN CODEC2_MODE_EN_DEFAULT
#endif
#if !defined(CODEC2_MODE_450_EN)
#define CODEC2_MODE_450_EN CODEC2_MODE_EN_DEFAULT
#endif
#if !defined(CODEC2_MODE_450PWB_EN)
#define CODEC2_MODE_450PWB_EN CODEC2_MODE_EN_DEFAULT
#endif
#define CODEC2_MODE_ACTIVE(mode_name, var) \
((mode_name##_EN) == 0 ? 0 : (var) == mode_name)

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@ -1,536 +0,0 @@
/*---------------------------------------------------------------------------*\
FILE........: newamp2.c
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
BASED ON....: "newamp1" by David Rowe
Quantisation functions for the sinusoidal coder, using "newamp1"
algorithm that resamples variable rate L [Am} to a fixed rate K then
VQs.
\*---------------------------------------------------------------------------*/
/*
Copyright David Rowe 2017
All rights reserved.
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License version 2.1, as
published by the Free Software Foundation. This program is
distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or
FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public
License for more details.
You should have received a copy of the GNU Lesser General Public License
along with this program; if not, see <http://www.gnu.org/licenses/>.
*/
#include "newamp2.h"
#include <assert.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "defines.h"
#include "mbest.h"
#include "newamp1.h"
#include "phase.h"
#include "quantise.h"
/*---------------------------------------------------------------------------*\
FUNCTION....: n2_mel_sample_freqs_kHz()
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
Outputs fixed frequencies for the K-Vectors to be able to work with both 8k
and 16k mode.
\*---------------------------------------------------------------------------*/
void n2_mel_sample_freqs_kHz(float rate_K_sample_freqs_kHz[], int K) {
float freq[] = {0.199816, 0.252849, 0.309008, 0.368476, 0.431449, 0.498134,
0.568749, 0.643526, 0.722710, 0.806561, 0.895354, 0.989380,
1.088948, 1.194384, 1.306034, 1.424264, 1.549463, 1.682041,
1.822432, 1.971098, 2.128525, 2.295232, 2.471763, 2.658699,
2.856652, 3.066272, 3.288246, 3.523303, 3.772214, 4.035795,
4.314912, 4.610478, 4.923465, 5.254899, 5.605865, 5.977518,
6.371075, 6.787827, 7.229141, 7.696465};
int k;
// printf("\n\n");
for (k = 0; k < K; k++) {
rate_K_sample_freqs_kHz[k] = freq[k];
// printf("%f ",mel);
// printf("%f \n",rate_K_sample_freqs_kHz[k]);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: n2_resample_const_rate_f() still equal to
resample_const_rate_f() AUTHOR......: David Rowe DATE CREATED: Jan 2017
Resample Am from time-varying rate L=floor(pi/Wo) to fixed rate K.
\*---------------------------------------------------------------------------*/
void n2_resample_const_rate_f(C2CONST *c2const, MODEL *model,
float rate_K_vec[],
float rate_K_sample_freqs_kHz[], int K) {
int m;
float AmdB[MAX_AMP + 1], rate_L_sample_freqs_kHz[MAX_AMP + 1], AmdB_peak;
/* convert rate L=pi/Wo amplitude samples to fixed rate K */
AmdB_peak = -100.0;
for (m = 1; m <= model->L; m++) {
AmdB[m] = 20.0 * log10(model->A[m] + 1E-16);
if (AmdB[m] > AmdB_peak) {
AmdB_peak = AmdB[m];
}
rate_L_sample_freqs_kHz[m] = m * model->Wo * (c2const->Fs / 2000.0) / M_PI;
// printf("m: %d AmdB: %f AmdB_peak: %f sf: %f\n", m, AmdB[m], AmdB_peak,
// rate_L_sample_freqs_kHz[m]);
}
/* clip between peak and peak -50dB, to reduce dynamic range */
for (m = 1; m <= model->L; m++) {
if (AmdB[m] < (AmdB_peak - 50.0)) {
AmdB[m] = AmdB_peak - 50.0;
}
}
interp_para(rate_K_vec, &rate_L_sample_freqs_kHz[1], &AmdB[1], model->L,
rate_K_sample_freqs_kHz, K);
}
/*---------------------------------------------------------------------------*\
FUNCTION....: n2_rate_K_mbest_encode
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
One stage rate K newamp2 VQ quantiser using mbest search.
\*---------------------------------------------------------------------------*/
void n2_rate_K_mbest_encode(int *indexes, float *x, float *xq, int ndim) {
int i, n1;
const float *codebook1 = newamp2vq_cb[0].cb;
struct MBEST *mbest_stage1;
float w[ndim];
int index[1];
/* codebook is compiled for a fixed K */
// assert(ndim == newamp2vq_cb[0].k);
/* equal weights, could be argued mel freq axis gives freq dep weighting */
for (i = 0; i < ndim; i++) w[i] = 1.0;
mbest_stage1 = mbest_create(1);
index[0] = 0;
/* Stage 1 */
mbest_search450(codebook1, x, w, ndim, NEWAMP2_K, newamp2vq_cb[0].m,
mbest_stage1, index);
n1 = mbest_stage1->list[0].index[0];
mbest_destroy(mbest_stage1);
// indexes[1]: legacy from newamp1
indexes[0] = n1;
indexes[1] = n1;
}
/*---------------------------------------------------------------------------*\
FUNCTION....: n2_resample_rate_L
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
Decoder side conversion of rate K vector back to rate L.
Plosives are set to zero for the first 2 of 4 frames.
\*---------------------------------------------------------------------------*/
void n2_resample_rate_L(C2CONST *c2const, MODEL *model, float rate_K_vec[],
float rate_K_sample_freqs_kHz[], int K,
int plosive_flag) {
float rate_K_vec_term[K + 2], rate_K_sample_freqs_kHz_term[K + 2];
float AmdB[MAX_AMP + 1], rate_L_sample_freqs_kHz[MAX_AMP + 1];
int m, k;
/* terminate either end of the rate K vecs with 0dB points */
rate_K_vec_term[0] = rate_K_vec_term[K + 1] = 0.0;
rate_K_sample_freqs_kHz_term[0] = 0.0;
rate_K_sample_freqs_kHz_term[K + 1] = 4.0;
for (k = 0; k < K; k++) {
rate_K_vec_term[k + 1] = rate_K_vec[k];
rate_K_sample_freqs_kHz_term[k + 1] = rate_K_sample_freqs_kHz[k];
// printf("k: %d f: %f rate_K: %f\n", k, rate_K_sample_freqs_kHz[k],
// rate_K_vec[k]);
}
for (m = 1; m <= model->L; m++) {
rate_L_sample_freqs_kHz[m] = m * model->Wo * (c2const->Fs / 2000.0) / M_PI;
}
interp_para(&AmdB[1], rate_K_sample_freqs_kHz_term, rate_K_vec_term, K + 2,
&rate_L_sample_freqs_kHz[1], model->L);
for (m = 1; m <= model->L; m++) {
if (plosive_flag == 0) {
model->A[m] = pow(10.0, AmdB[m] / 20.0);
} else {
model->A[m] = 0.1;
}
// printf("m: %d f: %f AdB: %f A: %f\n", m, rate_L_sample_freqs_kHz[m],
// AmdB[m], model->A[m]);
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: n2_post_filter_newamp2
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
Postfilter for the pseudo wideband mode. Still has to be adapted!
\*---------------------------------------------------------------------------*/
void n2_post_filter_newamp2(float vec[], float sample_freq_kHz[], int K,
float pf_gain) {
int k;
/*
vec is rate K vector describing spectrum of current frame lets
pre-emp before applying PF. 20dB/dec over 300Hz. Postfilter
affects energy of frame so we measure energy before and after
and normalise. Plenty of room for experiment here as well.
*/
float pre[K];
float e_before = 0.0;
float e_after = 0.0;
for (k = 0; k < K; k++) {
pre[k] = 20.0 * log10f(sample_freq_kHz[k] / 0.3);
vec[k] += pre[k];
e_before += POW10F(vec[k] / 10.0);
vec[k] *= pf_gain;
e_after += POW10F(vec[k] / 10.0);
}
float gain = e_after / e_before;
float gaindB = 10 * log10f(gain);
for (k = 0; k < K; k++) {
vec[k] -= gaindB;
vec[k] -= pre[k];
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: newamp2_model_to_indexes
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
newamp2 encoder: Encodes the 8k sampled samples using mbest search (one stage)
\*---------------------------------------------------------------------------*/
void newamp2_model_to_indexes(C2CONST *c2const, int indexes[], MODEL *model,
float rate_K_vec[],
float rate_K_sample_freqs_kHz[], int K,
float *mean, float rate_K_vec_no_mean[],
float rate_K_vec_no_mean_[], int plosive) {
int k;
/* convert variable rate L to fixed rate K */
resample_const_rate_f(c2const, model, rate_K_vec, rate_K_sample_freqs_kHz, K);
/* remove mean and two stage VQ */
float sum = 0.0;
for (k = 0; k < K; k++) sum += rate_K_vec[k];
*mean = sum / K;
for (k = 0; k < K; k++) {
rate_K_vec_no_mean[k] = rate_K_vec[k] - *mean;
}
// NEWAMP2_16K_K+1 because the last vector is not a vector for VQ (and not
// included in the constant) but a calculated medium mean value
n2_rate_K_mbest_encode(indexes, rate_K_vec_no_mean, rate_K_vec_no_mean_,
NEWAMP2_16K_K + 1);
/* scalar quantise mean (effectively the frame energy) */
float w[1] = {1.0};
float se;
indexes[2] = quantise(newamp2_energy_cb[0].cb, mean, w,
newamp2_energy_cb[0].k, newamp2_energy_cb[0].m, &se);
/* scalar quantise Wo. We steal the smallest Wo index to signal
an unvoiced frame */
if (model->voiced) {
int index = encode_log_Wo(c2const, model->Wo, 6);
if (index == 0) {
index = 1;
}
if (index == 63) {
index = 62;
}
indexes[3] = index;
} else {
indexes[3] = 0;
}
if (plosive != 0) {
indexes[3] = 63;
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: newamp2_indexes_to_rate_K_vec
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
newamp2 decoder for amplitudes {Am}. Given the rate K VQ and energy
indexes, outputs rate K vector. Equal to newamp1 but using only one stage VQ.
\*---------------------------------------------------------------------------*/
void newamp2_indexes_to_rate_K_vec(float rate_K_vec_[],
float rate_K_vec_no_mean_[],
float rate_K_sample_freqs_kHz[], int K,
float *mean_, int indexes[], float pf_gain) {
int k;
const float *codebook1 = newamp2vq_cb[0].cb;
int n1 = indexes[0];
for (k = 0; k < K; k++) {
rate_K_vec_no_mean_[k] = codebook1[(NEWAMP2_16K_K + 1) * n1 + k];
}
post_filter_newamp1(rate_K_vec_no_mean_, rate_K_sample_freqs_kHz, K, pf_gain);
*mean_ = newamp2_energy_cb[0].cb[indexes[2]];
for (k = 0; k < K; k++) {
rate_K_vec_[k] = rate_K_vec_no_mean_[k] + *mean_;
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: newamp2_16k_indexes_to_rate_K_vec
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
newamp2 decoder for amplitudes {Am}. Given the rate K VQ and energy
indexes, outputs rate K vector. Extends the sample rate by looking up the
corresponding higher frequency values with their energy difference to the base
energy (=>mean2)
\*---------------------------------------------------------------------------*/
void newamp2_16k_indexes_to_rate_K_vec(float rate_K_vec_[],
float rate_K_vec_no_mean_[],
float rate_K_sample_freqs_kHz[], int K,
float *mean_, int indexes[],
float pf_gain) {
int k;
const float *codebook1 = newamp2vq_cb[0].cb;
float mean2 = 0;
int n1 = indexes[0];
for (k = 0; k < K; k++) {
rate_K_vec_no_mean_[k] = codebook1[(K + 1) * n1 + k];
}
n2_post_filter_newamp2(rate_K_vec_no_mean_, rate_K_sample_freqs_kHz, K,
pf_gain);
*mean_ = newamp2_energy_cb[0].cb[indexes[2]];
mean2 = *mean_ + codebook1[(K + 1) * n1 + K] - 10;
// HF ear Protection
if (mean2 > 50) {
mean2 = 50;
}
for (k = 0; k < K; k++) {
if (k < NEWAMP2_K) {
rate_K_vec_[k] = rate_K_vec_no_mean_[k] + *mean_;
} else {
// Amplify or Reduce ??
rate_K_vec_[k] = rate_K_vec_no_mean_[k] + mean2;
}
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: newamp2_interpolate
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
Interpolates to the 4 10ms Frames and leaves the first 2 empty for plosives
\*---------------------------------------------------------------------------*/
void newamp2_interpolate(float interpolated_surface_[], float left_vec[],
float right_vec[], int K, int plosive_flag) {
int i, k;
int M = 4;
float c;
/* (linearly) interpolate 25Hz amplitude vectors back to 100Hz */
if (plosive_flag == 0) {
for (i = 0, c = 1.0; i < M; i++, c -= 1.0 / M) {
for (k = 0; k < K; k++) {
interpolated_surface_[i * K + k] =
left_vec[k] * c + right_vec[k] * (1.0 - c);
}
}
} else {
for (i = 0, c = 1.0; i < M; i++, c -= 1.0 / M) {
for (k = 0; k < K; k++) {
if (i < 2) {
interpolated_surface_[i * K + k] = 0;
} else {
// perhaps add some dB ?
interpolated_surface_[i * K + k] = right_vec[k];
}
}
}
}
}
/*---------------------------------------------------------------------------*\
FUNCTION....: newamp2_indexes_to_model
AUTHOR......: Thomas Kurin and Stefan Erhardt
INSTITUTE...: Institute for Electronics Engineering, University of
Erlangen-Nuremberg DATE CREATED: July 2018
newamp2 decoder. Chooses whether to decode to 16k mode or to 8k mode
\*---------------------------------------------------------------------------*/
void newamp2_indexes_to_model(C2CONST *c2const, MODEL model_[], COMP H[],
float *interpolated_surface_,
float prev_rate_K_vec_[], float *Wo_left,
int *voicing_left,
float rate_K_sample_freqs_kHz[], int K,
codec2_fft_cfg fwd_cfg, codec2_fft_cfg inv_cfg,
int indexes[], float pf_gain, int flag16k) {
float rate_K_vec_[K], rate_K_vec_no_mean_[K], mean_, Wo_right;
int voicing_right, k;
int M = 4;
/* extract latest rate K vector */
if (flag16k == 0) {
newamp2_indexes_to_rate_K_vec(rate_K_vec_, rate_K_vec_no_mean_,
rate_K_sample_freqs_kHz, K, &mean_, indexes,
pf_gain);
} else {
newamp2_16k_indexes_to_rate_K_vec(rate_K_vec_, rate_K_vec_no_mean_,
rate_K_sample_freqs_kHz, K, &mean_,
indexes, pf_gain);
}
/* decode latest Wo and voicing and plosive */
int plosive_flag = 0;
// Voiced with Wo
if (indexes[3] > 0 && indexes[3] < 63) {
Wo_right = decode_log_Wo(c2const, indexes[3], 6);
voicing_right = 1;
}
// Unvoiced
else if (indexes[3] == 0) {
Wo_right = 2.0 * M_PI / 100.0;
voicing_right = 0;
}
// indexes[3]=63 (= Plosive) and unvoiced
else {
Wo_right = 2.0 * M_PI / 100.0;
voicing_right = 0;
plosive_flag = 1;
}
/* interpolate 25Hz rate K vec back to 100Hz */
float *left_vec = prev_rate_K_vec_;
float *right_vec = rate_K_vec_;
newamp2_interpolate(interpolated_surface_, left_vec, right_vec, K,
plosive_flag);
/* interpolate 25Hz v and Wo back to 100Hz */
float aWo_[M];
int avoicing_[M], aL_[M], i;
interp_Wo_v(aWo_, aL_, avoicing_, *Wo_left, Wo_right, *voicing_left,
voicing_right);
/* back to rate L amplitudes, synthesis phase for each frame */
for (i = 0; i < M; i++) {
model_[i].Wo = aWo_[i];
model_[i].L = aL_[i];
model_[i].voiced = avoicing_[i];
// Plosive Detected
if (plosive_flag > 0) {
// First two frames are set to zero
if (i < 2) {
n2_resample_rate_L(c2const, &model_[i], &interpolated_surface_[K * i],
rate_K_sample_freqs_kHz, K, 1);
} else {
n2_resample_rate_L(c2const, &model_[i], &interpolated_surface_[K * i],
rate_K_sample_freqs_kHz, K, 0);
}
}
// No Plosive, standard resample
else {
n2_resample_rate_L(c2const, &model_[i], &interpolated_surface_[K * i],
rate_K_sample_freqs_kHz, K, 0);
}
determine_phase(c2const, &H[(MAX_AMP + 1) * i], &model_[i],
NEWAMP2_PHASE_NFFT, fwd_cfg, inv_cfg);
}
/* update memories for next time */
for (k = 0; k < K; k++) {
prev_rate_K_vec_[k] = rate_K_vec_[k];
}
*Wo_left = Wo_right;
*voicing_left = voicing_right;
}