static void sbr_autocorrelate_c(const float x[40][2], float phi[3][2][2])
{
#if 0
/* This code is slower because it multiplies memory accesses.
* It is left for educational purposes and because it may offer
* a better reference for writing arch-specific DSP functions. */
autocorrelate(x, phi, 0);
autocorrelate(x, phi, 1);
autocorrelate(x, phi, 2);
#else
float real_sum2 = x[0][0] * x[2][0] + x[0][1] * x[2][1];
float imag_sum2 = x[0][0] * x[2][1] - x[0][1] * x[2][0];
float real_sum1 = 0.0f, imag_sum1 = 0.0f, real_sum0 = 0.0f;
int i;
for (i = 1; i < 38; i++) {
real_sum0 += x[i][0] * x[i ][0] + x[i][1] * x[i ][1];
real_sum1 += x[i][0] * x[i + 1][0] + x[i][1] * x[i + 1][1];
imag_sum1 += x[i][0] * x[i + 1][1] - x[i][1] * x[i + 1][0];
real_sum2 += x[i][0] * x[i + 2][0] + x[i][1] * x[i + 2][1];
imag_sum2 += x[i][0] * x[i + 2][1] - x[i][1] * x[i + 2][0];
}
phi[2 - 2][1][0] = real_sum2;
phi[2 - 2][1][1] = imag_sum2;
phi[2 ][1][0] = real_sum0 + x[ 0][0] * x[ 0][0] + x[ 0][1] * x[ 0][1];
phi[1 ][0][0] = real_sum0 + x[38][0] * x[38][0] + x[38][1] * x[38][1];
phi[2 - 1][1][0] = real_sum1 + x[ 0][0] * x[ 1][0] + x[ 0][1] * x[ 1][1];
phi[2 - 1][1][1] = imag_sum1 + x[ 0][0] * x[ 1][1] - x[ 0][1] * x[ 1][0];
phi[0 ][0][0] = real_sum1 + x[38][0] * x[39][0] + x[38][1] * x[39][1];
phi[0 ][0][1] = imag_sum1 + x[38][0] * x[39][1] - x[38][1] * x[39][0];
#endif
}
void find_aks_for_lsp(
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[N]; /* windowed frame of Nsam speech samples */
float R[P+1]; /* order+1 autocorrelation values of Sn[] */
int i;
hanning_window(Sn,Wn,Nsam);
autocorrelate(Wn,R,Nsam,order);
R[0] += LPC_FLOOR;
assert(order == 10); /* lag window only defined for order == 10 */
for(i=0; i<=order; i++)
R[i] *= lag_window[i];
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;
}
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;
}
void find_aks(
scalar Sn[], /* Nsam samples with order sample memory */
scalar a[], /* order+1 LPCs with first coeff 1.0 */
int Nsam, /* number of input speech samples */
int order, /* order of the LPC analysis */
scalar *E /* residual energy */
)
{
scalar Wn[LPC_MAX_N]; /* windowed frame of Nsam speech samples */
scalar R[order+1]; /* order+1 autocorrelation values of Sn[] */
int i;
assert(Nsam < LPC_MAX_N);
hanning_window(Sn,Wn,Nsam);
autocorrelate(Wn,R,Nsam,order);
levinson_durbin(R,a,order);
*E = fl_to_numb(0.0);
for(i=0; i<=order; i++)
*E = s_add(*E , s_mul(a[i],R[i]));
if (*E < fl_to_numb(0.0)) {
#ifdef MATH_Q16_16
*E = 1;
#else
*E = powf(2, -16);//fl_to_numb(1E-12); For debuging purposes.
#endif
}
}
float speech_to_uq_lsps(float lsp[],
float ak[],
float Sn[],
float w[],
int order
)
{
int i, roots;
float Wn[M];
float R[LPC_MAX+1];
float e, E;
e = 0.0;
for(i=0; i<M; i++) {
Wn[i] = Sn[i]*w[i];
e += Wn[i]*Wn[i];
}
/* trap 0 energy case as LPC analysis will fail */
if (e == 0.0) {
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
return 0.0;
}
autocorrelate(Wn, R, M, order);
levinson_durbin(R, ak, order);
E = 0.0;
for(i=0; i<=order; i++)
E += ak[i]*R[i];
/* 15 Hz BW expansion as I can't hear the difference and it may help
help occasional fails in the LSP root finding. Important to do this
after energy calculation to avoid -ve energy values.
*/
for(i=0; i<=order; i++)
ak[i] *= powf(0.994,(float)i);
roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
if (roots != order) {
/* if root finding fails use some benign LSP values instead */
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
}
return E;
}
void Histogram::compute() {
counts.resize( ceil( bin_size ) + 1 );
std::fill( counts.begin(), counts.end(), 0 );
Bins::Position low_border = Bins::Position::Constant( 0 * camera::pixel ),
high_border = bins.sizes().array() - 1 * camera::pixel;
for ( Bins::iterator i = bins.begin(); i != bins.end(); ++i ) {
if ( (i.position() == low_border).any() ||
(i.position() == high_border).any() )
continue;
autocorrelate( *i );
for (ForwardScan::iterator j = forward_scan.begin(); j != forward_scan.end(); ++j) {
crosscorrelate( *i, bins( i.position() + *j ) );
}
}
if ( periodic_boundary )
counts.pop_back();
}
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);
if (roots != order) {
/* for some reason LSP roots could not be found */
/* some alpha testers are reporting this condition */
fprintf(stderr, "LSP roots not found!\nroots = %d\n", roots);
for(i=0; i<=order; i++)
fprintf(stderr, "a[%d] = %f\n", i, ak[i]);
/* some benign LSP values we can use instead */
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
}
return E;
}
static void sbr_autocorrelate_c(const float x[40][2], float phi[3][2][2])
{
autocorrelate(x, phi, 0);
autocorrelate(x, phi, 1);
autocorrelate(x, phi, 2);
}
int main(int argc, char *argv[])
{
FILE *fout = NULL; /* output speech file */
FILE *fin; /* input speech file */
short buf[N]; /* input/output buffer */
float buf_float[N];
float Sn[M]; /* float input speech samples */
float Sn_pre[N]; /* pre-emphasised input speech samples */
COMP Sw[FFT_ENC]; /* DFT of Sn[] */
kiss_fft_cfg fft_fwd_cfg;
kiss_fft_cfg fft_inv_cfg;
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,m; /* loop variable */
int frames;
float prev_Wo, prev__Wo, prev_uq_Wo;
float pitch;
char out_file[MAX_STR];
char ampexp_arg[MAX_STR];
char phaseexp_arg[MAX_STR];
float snr;
float sum_snr;
int orderi;
int lpc_model = 0, order = LPC_ORD;
int lsp = 0, lspd = 0, lspvq = 0;
int lspres = 0;
int lspjvm = 0, lspjnd = 0, lspmel = 0;
#ifdef __EXPERIMENTAL__
int lspanssi = 0,
#endif
int prede = 0;
float pre_mem = 0.0, de_mem = 0.0;
float ak[order];
COMP Sw_[FFT_ENC];
COMP Ew[FFT_ENC];
int phase0 = 0;
float ex_phase[MAX_AMP+1];
int postfilt;
int hand_voicing = 0, phaseexp = 0, ampexp = 0, hi = 0, simlpcpf = 0;
int lpcpf = 0;
FILE *fvoicing = 0;
MODEL prev_model;
int dec;
int decimate = 1;
float lsps[order];
float e, prev_e;
int lsp_indexes[order];
float lsps_[order];
float Woe_[2];
float lsps_dec[4][LPC_ORD], e_dec[4], weight, weight_inc, ak_dec[4][LPC_ORD];
MODEL model_dec[4], prev_model_dec;
float prev_lsps_dec[order], prev_e_dec;
void *nlp_states;
float hpf_states[2];
int scalar_quant_Wo_e = 0;
int vector_quant_Wo_e = 0;
int dump_pitch_e = 0;
FILE *fjvm = NULL;
#ifdef DUMP
int dump;
#endif
struct PEXP *pexp = NULL;
struct AEXP *aexp = NULL;
float gain = 1.0;
int bpf_en = 0;
float bpf_buf[BPF_N+N];
char* opt_string = "ho:";
struct option long_options[] = {
{ "lpc", required_argument, &lpc_model, 1 },
{ "lspjnd", no_argument, &lspjnd, 1 },
{ "lspmel", no_argument, &lspmel, 1 },
{ "lsp", no_argument, &lsp, 1 },
{ "lspd", no_argument, &lspd, 1 },
{ "lspvq", no_argument, &lspvq, 1 },
{ "lspres", no_argument, &lspres, 1 },
{ "lspjvm", no_argument, &lspjvm, 1 },
#ifdef __EXPERIMENTAL__
{ "lspanssi", no_argument, &lspanssi, 1 },
#endif
{ "phase0", no_argument, &phase0, 1 },
{ "phaseexp", required_argument, &phaseexp, 1 },
{ "ampexp", required_argument, &exp, 1 },
{ "postfilter", no_argument, &postfilt, 1 },
{ "hand_voicing", required_argument, &hand_voicing, 1 },
{ "dec", required_argument, &dec, 1 },
{ "hi", no_argument, &hi, 1 },
{ "simlpcpf", no_argument, &simlpcpf, 1 },
{ "lpcpf", no_argument, &lpcpf, 1 },
{ "prede", no_argument, &prede, 1 },
{ "dump_pitch_e", required_argument, &dump_pitch_e, 1 },
{ "sq_pitch_e", no_argument, &scalar_quant_Wo_e, 1 },
{ "vq_pitch_e", no_argument, &vector_quant_Wo_e, 1 },
{ "rate", required_argument, NULL, 0 },
{ "gain", required_argument, NULL, 0 },
{ "bpf", no_argument, &bpf_en, 1 },
#ifdef DUMP
{ "dump", required_argument, &dump, 1 },
#endif
{ "help", no_argument, NULL, 'h' },
{ NULL, no_argument, NULL, 0 }
};
int num_opts=sizeof(long_options)/sizeof(struct option);
COMP Aw[FFT_ENC];
for(i=0; i<M; i++) {
Sn[i] = 1.0;
Sn_pre[i] = 1.0;
}
for(i=0; i<2*N; i++)
Sn_[i] = 0;
prev_uq_Wo = prev_Wo = 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] = (PI/3)*(float)rand()/RAND_MAX;
ex_phase[i] = 0.0;
}
e = prev_e = 1;
hpf_states[0] = hpf_states[1] = 0.0;
nlp_states = nlp_create(M);
if (argc < 2) {
print_help(long_options, num_opts, argv);
}
/*----------------------------------------------------------------*\
Interpret Command Line Arguments
\*----------------------------------------------------------------*/
while(1) {
int option_index = 0;
int opt = getopt_long(argc, argv, opt_string,
long_options, &option_index);
if (opt == -1)
break;
switch (opt) {
case 0:
if(strcmp(long_options[option_index].name, "lpc") == 0) {
orderi = atoi(optarg);
if((orderi < 4) || (orderi > order)) {
fprintf(stderr, "Error in LPC order (4 to %d): %s\n", order, optarg);
exit(1);
}
order = orderi;
#ifdef DUMP
} else if(strcmp(long_options[option_index].name, "dump") == 0) {
if (dump)
dump_on(optarg);
#endif
} else if(strcmp(long_options[option_index].name, "lsp") == 0
|| strcmp(long_options[option_index].name, "lspd") == 0
|| strcmp(long_options[option_index].name, "lspvq") == 0) {
assert(order == LPC_ORD);
} else if(strcmp(long_options[option_index].name, "dec") == 0) {
decimate = atoi(optarg);
if ((decimate != 2) && (decimate != 4)) {
fprintf(stderr, "Error in --dec, must be 2 or 4\n");
exit(1);
}
if (!phase0) {
printf("needs --phase0 to resample phase when using --dec\n");
exit(1);
}
if (!lpc_model) {
printf("needs --lpc [order] to resample amplitudes when using --dec\n");
exit(1);
}
} else if(strcmp(long_options[option_index].name, "hand_voicing") == 0) {
if ((fvoicing = fopen(optarg,"rt")) == NULL) {
fprintf(stderr, "Error opening voicing file: %s: %s.\n",
optarg, strerror(errno));
exit(1);
}
} else if(strcmp(long_options[option_index].name, "dump_pitch_e") == 0) {
if ((fjvm = fopen(optarg,"wt")) == NULL) {
fprintf(stderr, "Error opening pitch & energy dump file: %s: %s.\n",
optarg, strerror(errno));
exit(1);
}
} else if(strcmp(long_options[option_index].name, "phaseexp") == 0) {
strcpy(phaseexp_arg, optarg);
} else if(strcmp(long_options[option_index].name, "ampexp") == 0) {
strcpy(ampexp_arg, optarg);
} else if(strcmp(long_options[option_index].name, "gain") == 0) {
gain = atof(optarg);
} else if(strcmp(long_options[option_index].name, "rate") == 0) {
if(strcmp(optarg,"3200") == 0) {
lpc_model = 1;
scalar_quant_Wo_e = 1;
lspd = 1;
phase0 = 1;
postfilt = 1;
decimate = 1;
lpcpf = 1;
} else if(strcmp(optarg,"2400") == 0) {
lpc_model = 1;
vector_quant_Wo_e = 1;
lsp = 1;
phase0 = 1;
postfilt = 1;
decimate = 2;
lpcpf = 1;
} else if(strcmp(optarg,"1400") == 0) {
lpc_model = 1;
vector_quant_Wo_e = 1;
lsp = 1;
phase0 = 1;
postfilt = 1;
decimate = 4;
lpcpf = 1;
} else if(strcmp(optarg,"1300") == 0) {
lpc_model = 1;
scalar_quant_Wo_e = 1;
lsp = 1;
phase0 = 1;
postfilt = 1;
decimate = 4;
lpcpf = 1;
} else if(strcmp(optarg,"1200") == 0) {
lpc_model = 1;
scalar_quant_Wo_e = 1;
lspjvm = 1;
phase0 = 1;
postfilt = 1;
decimate = 4;
lpcpf = 1;
} else {
fprintf(stderr, "Error: invalid output rate (3200|2400|1400|1200) %s\n", optarg);
exit(1);
}
}
break;
case 'h':
print_help(long_options, num_opts, argv);
break;
case 'o':
if (strcmp(optarg, "-") == 0) fout = stdout;
else if ((fout = fopen(optarg,"wb")) == NULL) {
fprintf(stderr, "Error opening output speech file: %s: %s.\n",
optarg, strerror(errno));
exit(1);
}
strcpy(out_file,optarg);
break;
default:
/* This will never be reached */
break;
}
}
/* Input file */
if (strcmp(argv[optind], "-") == 0) fin = stdin;
else if ((fin = fopen(argv[optind],"rb")) == NULL) {
fprintf(stderr, "Error opening input speech file: %s: %s.\n",
argv[optind], strerror(errno));
exit(1);
}
ex_phase[0] = 0;
Woe_[0] = Woe_[1] = 1.0;
/*
printf("lspd: %d lspdt: %d lspdt_mode: %d phase0: %d postfilt: %d "
"decimate: %d dt: %d\n",lspd,lspdt,lspdt_mode,phase0,postfilt,
decimate,dt);
*/
/* Initialise ------------------------------------------------------------*/
fft_fwd_cfg = kiss_fft_alloc(FFT_ENC, 0, NULL, NULL); /* fwd FFT,used in several places */
fft_inv_cfg = kiss_fft_alloc(FFT_DEC, 1, NULL, NULL); /* inverse FFT, used just for synth */
make_analysis_window(fft_fwd_cfg, w, W);
make_synthesis_window(Pn);
quantise_init();
if (phaseexp)
pexp = phase_experiment_create();
if (ampexp)
aexp = amp_experiment_create();
if (bpf_en) {
for(i=0; i<BPF_N; i++)
bpf_buf[i] = 0.0;
}
for(i=0; i<LPC_ORD; i++) {
prev_lsps_dec[i] = i*PI/(LPC_ORD+1);
}
prev_e_dec = 1;
for(m=1; m<=MAX_AMP; m++)
prev_model_dec.A[m] = 0.0;
prev_model_dec.Wo = TWO_PI/P_MAX;
prev_model_dec.L = PI/prev_model_dec.Wo;
prev_model_dec.voiced = 0;
/*----------------------------------------------------------------* \
Main Loop
\*----------------------------------------------------------------*/
frames = 0;
sum_snr = 0;
while(fread(buf,sizeof(short),N,fin)) {
frames++;
for(i=0; i<N; i++)
buf_float[i] = buf[i];
/* optionally filter input speech */
if (prede) {
pre_emp(Sn_pre, buf_float, &pre_mem, N);
for(i=0; i<N; i++)
buf_float[i] = Sn_pre[i];
}
if (bpf_en) {
for(i=0; i<BPF_N; i++)
bpf_buf[i] = bpf_buf[N+i];
for(i=0; i<N; i++)
bpf_buf[BPF_N+i] = buf_float[i];
inverse_filter(&bpf_buf[BPF_N], bpf, N, buf_float, BPF_N);
}
/* shift buffer of input samples, and insert new samples */
for(i=0; i<M-N; i++) {
Sn[i] = Sn[i+N];
}
for(i=0; i<N; i++)
Sn[i+M-N] = buf_float[i];
/*------------------------------------------------------------*\
Estimate Sinusoidal Model Parameters
\*------------------------------------------------------------*/
nlp(nlp_states,Sn,N,P_MIN,P_MAX,&pitch,Sw,W,&prev_uq_Wo);
model.Wo = TWO_PI/pitch;
dft_speech(fft_fwd_cfg, Sw, Sn, w);
two_stage_pitch_refinement(&model, Sw);
estimate_amplitudes(&model, Sw, W, 1);
#ifdef DUMP
dump_Sn(Sn); dump_Sw(Sw); dump_model(&model);
#endif
if (ampexp)
amp_experiment(aexp, &model, ampexp_arg);
if (phaseexp) {
#ifdef DUMP
dump_phase(&model.phi[0], model.L);
#endif
phase_experiment(pexp, &model, phaseexp_arg);
#ifdef DUMP
dump_phase_(&model.phi[0], model.L);
#endif
}
if (hi) {
int m;
for(m=1; m<model.L/2; m++)
model.A[m] = 0.0;
for(m=3*model.L/4; m<=model.L; m++)
model.A[m] = 0.0;
}
/*------------------------------------------------------------*\
Zero-phase modelling
\*------------------------------------------------------------*/
if (phase0) {
float Wn[M]; /* windowed speech samples */
float Rk[order+1]; /* autocorrelation coeffs */
COMP a[FFT_ENC];
#ifdef DUMP
dump_phase(&model.phi[0], model.L);
#endif
/* 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,order);
levinson_durbin(Rk,ak,order);
/* determine voicing */
snr = est_voicing_mbe(&model, Sw, W, Sw_, Ew);
if (dump_pitch_e)
fprintf(fjvm, "%f %f %d ", model.Wo, snr, model.voiced);
//printf("snr %3.2f v: %d Wo: %f prev_Wo: %f\n", snr, model.voiced,
// model.Wo, prev_uq_Wo);
#ifdef DUMP
dump_Sw_(Sw_);
dump_Ew(Ew);
dump_snr(snr);
#endif
/* just to make sure we are not cheating - kill all phases */
for(i=0; i<=MAX_AMP; i++)
model.phi[i] = 0;
/* Determine DFT of A(exp(jw)), which is needed for phase0 model when
LPC is not used, e.g. indecimate=1 (10ms) frames with no LPC */
for(i=0; i<FFT_ENC; i++) {
a[i].real = 0.0;
a[i].imag = 0.0;
}
for(i=0; i<=order; i++)
a[i].real = ak[i];
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)a, (kiss_fft_cpx *)Aw);
if (hand_voicing) {
fscanf(fvoicing,"%d\n",&model.voiced);
}
}
/*------------------------------------------------------------*\
LPC model amplitudes and LSP quantisation
\*------------------------------------------------------------*/
if (lpc_model) {
e = speech_to_uq_lsps(lsps, ak, Sn, w, order);
for(i=0; i<LPC_ORD; i++)
lsps_[i] = lsps[i];
#ifdef DUMP
dump_ak(ak, order);
dump_E(e);
#endif
/* tracking down -ve energy values with BW expansion */
/*
if (e < 0.0) {
int i;
FILE*f=fopen("x.txt","wt");
for(i=0; i<M; i++)
fprintf(f,"%f\n", Sn[i]);
fclose(f);
printf("e = %f frames = %d\n", e, frames);
for(i=0; i<order; i++)
printf("%f ", ak[i]);
exit(0);
}
*/
if (dump_pitch_e)
fprintf(fjvm, "%f\n", e);
#ifdef DUMP
dump_lsp(lsps);
#endif
/* various LSP quantisation schemes */
if (lsp) {
encode_lsps_scalar(lsp_indexes, lsps, LPC_ORD);
decode_lsps_scalar(lsps_, lsp_indexes, LPC_ORD);
bw_expand_lsps(lsps_, LPC_ORD, 50.0, 100.0);
lsp_to_lpc(lsps_, ak, LPC_ORD);
}
if (lspd) {
encode_lspds_scalar(lsp_indexes, lsps, LPC_ORD);
decode_lspds_scalar(lsps_, lsp_indexes, LPC_ORD);
lsp_to_lpc(lsps_, ak, LPC_ORD);
}
#ifdef __EXPERIMENTAL__
if (lspvq) {
lspvq_quantise(lsps, lsps_, LPC_ORD);
bw_expand_lsps(lsps_, LPC_ORD, 50.0, 100.0);
lsp_to_lpc(lsps_, ak, LPC_ORD);
}
#endif
if (lspjvm) {
/* Jean-Marc's multi-stage, split VQ */
lspjvm_quantise(lsps, lsps_, LPC_ORD);
{
float lsps_bw[LPC_ORD];
memcpy(lsps_bw, lsps_, sizeof(float)*LPC_ORD);
bw_expand_lsps(lsps_bw, LPC_ORD, 50.0, 100.0);
lsp_to_lpc(lsps_bw, ak, LPC_ORD);
}
}
#ifdef __EXPERIMENTAL__
if (lspanssi) {
/* multi-stage VQ from Anssi Ramo OH3GDD */
lspanssi_quantise(lsps, lsps_, LPC_ORD, 5);
bw_expand_lsps(lsps_, LPC_ORD, 50.0, 100.0);
lsp_to_lpc(lsps_, ak, LPC_ORD);
}
#endif
/* experimenting with non-linear LSP spacing to see if
it's just noticable */
if (lspjnd) {
for(i=0; i<LPC_ORD; i++)
lsps_[i] = lsps[i];
locate_lsps_jnd_steps(lsps_, LPC_ORD);
lsp_to_lpc(lsps_, ak, LPC_ORD);
}
/* Another experiment with non-linear LSP spacing, this
time using a scaled version of mel frequency axis
warping. The scaling is such that the integer output
can be directly sent over the channel.
*/
if (lspmel) {
float f, f_;
float mel[LPC_ORD];
int mel_indexes[LPC_ORD];
for(i=0; i<order; i++) {
f = (4000.0/PI)*lsps[i];
mel[i] = floor(2595.0*log10(1.0 + f/700.0) + 0.5);
}
for(i=1; i<order; i++) {
if (mel[i] == mel[i-1])
mel[i]++;
}
encode_mels_scalar(mel_indexes, mel, 6);
decode_mels_scalar(mel, mel_indexes, 6);
#ifdef DUMP
dump_mel(mel, order);
#endif
for(i=0; i<LPC_ORD; i++) {
f_ = 700.0*( pow(10.0, (float)mel[i]/2595.0) - 1.0);
lsps_[i] = f_*(PI/4000.0);
}
lsp_to_lpc(lsps_, ak, order);
}
if (scalar_quant_Wo_e) {
e = decode_energy(encode_energy(e, E_BITS), E_BITS);
model.Wo = decode_Wo(encode_Wo(model.Wo, WO_BITS), WO_BITS);
model.L = PI/model.Wo; /* if we quantise Wo re-compute L */
}
if (vector_quant_Wo_e) {
/* JVM's experimental joint Wo & LPC energy quantiser */
quantise_WoE(&model, &e, Woe_);
}
}
/*------------------------------------------------------------*\
Synthesise and optional decimation to 20 or 40ms frame rate
\*------------------------------------------------------------*/
/*
if decimate == 2, we interpolate frame n from frame n-1 and n+1
if decimate == 4, we interpolate frames n, n+1, n+2, from frames n-1 and n+3
This is meant to give identical results to the implementations of various modes
in codec2.c
*/
/* delay line to keep frame by frame voicing decisions */
for(i=0; i<decimate-1; i++)
model_dec[i] = model_dec[i+1];
model_dec[decimate-1] = model;
if ((frames % decimate) == 0) {
for(i=0; i<order; i++)
lsps_dec[decimate-1][i] = lsps_[i];
e_dec[decimate-1] = e;
model_dec[decimate-1] = model;
/* interpolate the model parameters */
weight_inc = 1.0/decimate;
for(i=0, weight=weight_inc; i<decimate-1; i++, weight += weight_inc) {
//model_dec[i].voiced = model_dec[decimate-1].voiced;
interpolate_lsp_ver2(&lsps_dec[i][0], prev_lsps_dec, &lsps_dec[decimate-1][0], weight, order);
interp_Wo2(&model_dec[i], &prev_model_dec, &model_dec[decimate-1], weight);
e_dec[i] = interp_energy2(prev_e_dec, e_dec[decimate-1],weight);
}
/* then recover spectral amplitudes and synthesise */
for(i=0; i<decimate; i++) {
if (lpc_model) {
lsp_to_lpc(&lsps_dec[i][0], &ak_dec[i][0], order);
aks_to_M2(fft_fwd_cfg, &ak_dec[i][0], order, &model_dec[i], e_dec[i],
&snr, 0, simlpcpf, lpcpf, 1, LPCPF_BETA, LPCPF_GAMMA, Aw);
apply_lpc_correction(&model_dec[i]);
#ifdef DUMP
dump_lsp_(&lsps_dec[i][0]);
dump_ak_(&ak_dec[i][0], order);
sum_snr += snr;
dump_quantised_model(&model_dec[i]);
#endif
}
if (phase0)
phase_synth_zero_order(fft_fwd_cfg, &model_dec[i], ex_phase, Aw);
synth_one_frame(fft_inv_cfg, buf, &model_dec[i], Sn_, Pn, prede, &de_mem, gain);
if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
}
/*
for(i=0; i<decimate; i++) {
printf("%d Wo: %f L: %d v: %d\n", frames, model_dec[i].Wo, model_dec[i].L, model_dec[i].voiced);
}
if (frames == 4*50)
exit(0);
*/
/* update memories for next frame ----------------------------*/
prev_model_dec = model_dec[decimate-1];
prev_e_dec = e_dec[decimate-1];
for(i=0; i<LPC_ORD; i++)
prev_lsps_dec[i] = lsps_dec[decimate-1][i];
}
}
/*----------------------------------------------------------------*\
End Main Loop
\*----------------------------------------------------------------*/
fclose(fin);
if (fout != NULL)
fclose(fout);
if (lpc_model)
printf("SNR av = %5.2f dB\n", sum_snr/frames);
if (phaseexp)
phase_experiment_destroy(pexp);
if (ampexp)
amp_experiment_destroy(aexp);
#ifdef DUMP
if (dump)
dump_off();
#endif
if (hand_voicing)
fclose(fvoicing);
nlp_destroy(nlp_states);
return 0;
}
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;
const 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_cb[i].k;
m = lsp_cb[i].m;
cb = lsp_cb[i].cb;
index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
lsp_hz[i] = cb[index*k];
}
/* 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);
#ifdef DUMP
dump_lsp(lsp);
#endif
}
#ifdef DUMP
dump_E(E);
#endif
#ifdef SIM_QUANT
/* simulated LPC energy quantisation */
{
float e = 10.0*log10(E);
e += 2.0*(1.0 - 2.0*(float)rand()/RAND_MAX);
E = pow(10.0,e/10.0);
}
#endif
aks_to_M2(ak,order,model,E,&snr, 1); /* {ak} -> {Am} LPC decode */
return snr;
}
double PitchDetector::detectAcfPitchForBlock (float* samples, int numSamples)
{
const int minSample = int (sampleRate / maxFrequency);
const int maxSample = int (sampleRate / minFrequency);
lowFilter.reset();
highFilter.reset();
lowFilter.processSamples (samples, numSamples);
highFilter.processSamples (samples, numSamples);
autocorrelate (samples, numSamples, buffer1.getData());
normalise (buffer1.getData(), buffer1.getSize());
// float max = 0.0f;
// int sampleIndex = 0;
// for (int i = minSample; i < maxSample; ++i)
// {
// const float sample = buffer1.getData()[i];
// if (sample > max)
// {
// max = sample;
// sampleIndex = i;
// }
// }
float* bufferData = buffer1.getData();
// const int bufferSize = buffer1.getSize();
int firstNegativeZero = 0;
// first peak method
for (int i = 0; i < numSamples - 1; ++i)
{
if (bufferData[i] >= 0.0f && bufferData[i + 1] < 0.0f)
{
firstNegativeZero = i;
break;
}
}
// apply gain ramp
// float rampDelta = 1.0f / numSamples;
// float rampLevel = 1.0f;
// for (int i = 0; i < numSamples - 1; ++i)
// {
// bufferData[i] *= cubeNumber (rampLevel);
// rampLevel -= rampDelta;
// }
float max = -1.0f;
int sampleIndex = 0;
for (int i = jmax (firstNegativeZero, minSample); i < maxSample; ++i)
{
if (bufferData[i] > max)
{
max = bufferData[i];
sampleIndex = i;
}
}
// buffer2.setSizeQuick (numSamples);
/* autocorrelate (buffer1.getData(), buffer1.getSize(), buffer2.getData());
normalise (buffer2.getData(), buffer2.getSize());*/
//buffer2.quickCopy (buffer1.getData(), buffer1.getSize());
// differentiate (buffer1.getData(), buffer1.getSize(), buffer2.getData());
// normalise (buffer2.getData()+2, buffer2.getSize()-2);
// differentiate (buffer2.getData(), buffer2.getSize(), buffer2.getData());
/* for (int i = minSample + 1; i < maxSample - 1; ++i)
{
const float previousSample = buffer2.getData()[i - 1];
const float sample = buffer2.getData()[i];
const float nextSample = buffer2.getData()[i + 1];
if (sample > previousSample
&& sample > nextSample
&& sample > 0.5f)
sampleIndex = i;
}*/
//differentiate (buffer2.getData(), buffer2.getSize(), buffer2.getData());
//normalise (buffer2.getData() + minSample, buffer2.getSize() - minSample);
// float min = 0.0f;
// int sampleIndex = 0;
// for (int i = minSample; i < maxSample; ++i)
// {
// const float sample = buffer2.getData()[i];
// if (sample < min)
// {
// min = sample;
// sampleIndex = i;
// }
// }
if (sampleIndex > 0)
return sampleRate / sampleIndex;
else
return 0.0;
}
main(int argc, char *argv[])
{
FILE *fp;
int fd,arg;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int rate=8000;
float f[WINDOW],hann[WINDOW],w[WINDOW],w2[WINDOW],s0,s1=0,tot;
float ac[ORDER+1],lcp[ORDER+1],lsp[ORDER],l[ORDER],weight[ORDER],delta,d;
short sample,s[160],buf[2000];
int i,j,n,b,toggle=1;
float e,laste=0;
int ebit=ETOPBIT, ebuff=0;
int sound=0; // boolean start/stop
float f2[FFT],min;
float real[FFT],imag[FFT];
int dummy[100000];
float amp[WINDOW],pha[WINDOW];
int frame=0;
SpeexPreprocessState *st;
for (i=0; i<8; i++) report[i]=0;
st = speex_preprocess_state_init(160, 8000);
i=1;
speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_DENOISE, &i);
// i=1;
// speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_DEREVERB, &i);
// e=.0;
// speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_DEREVERB_DECAY, &e);
// e=.0;
// speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL, &e);
setup_twiddles(realtwiddle,imtwiddle,FFT);
for (i=0; i<WINDOW; i++) f[i]=0;
for (i=0; i<ORDER; i++) {last[i]=0; last2[i]=0;}
for(i=0; i<WINDOW; i++) hann[i]=0.5-0.5*cos(2.0*M_PI*i/(WINDOW-1));
if (argc==2) training=atoi(argv[1]);
fprintf(stderr,"training=%i\n",training); // exit(0);
cbsize=0; start[0]=0; size[0]=0;
if (training==0) if (fp=fopen("cb.txt","r")) {
while(!feof(fp)) {
fscanf(fp,"%i\n",&size[cbsize]);
for (i=start[cbsize]; i<start[cbsize]+size[cbsize]; i++) {
for (n=1; n<FF2-1; n++) fscanf(fp,"%f,",&cb[i][n]); fscanf(fp,"\n");
}
start[cbsize+1]=start[cbsize]+size[cbsize]; cbsize++;
}
fclose(fp);
}
//for (i=0; i<cbsize; i++) printf("%i,\n",size[i]); exit(0);
//---------------------------------
fp=fopen("/tmp/b.raw","w");
snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0);
snd_pcm_hw_params_malloc(&hw_params);
snd_pcm_hw_params_any(handle, hw_params);
snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate_near(handle, hw_params, &rate, 0);
snd_pcm_hw_params_set_channels(handle, hw_params, 2);
snd_pcm_hw_params(handle, hw_params);
snd_pcm_hw_params_free(hw_params);
snd_pcm_prepare(handle);
//printf("sleep 1...\n"); sleep(1); printf("OK, go....\n");
while(1) {
for (i=0; i<WINDOW-STEP; i++) f[i]=f[i+STEP]; // shift samples down
if (toggle) {
//read(fd,s,160*2);
snd_pcm_readi(handle, buf, 160);
for (i=0; i<160; i++) s[i]=buf[i*2];
speex_preprocess_run(st, s);
}
else bcopy(&s[80],s,80*2);
toggle=!toggle;
for (i=WINDOW-STEP,j=0; i<WINDOW; i++,j++) {
sample=s[j]; s0=(float)sample;
f[i]=s0-s1*EMPH; s1=s0; // 1.0 pre-emphasis
fwrite(&sample,2,1,fp);
}
for (i=0; i<WINDOW; i++) w[i]=f[i];
// remove any DC level....
tot=0; for (i=0; i<WINDOW; i++) tot+=w[i]; tot/=WINDOW;
for (i=0; i<WINDOW; i++) w[i]-=tot;
for (i=0; i<WINDOW; i++) w[i]*=hann[i]; // window data
autocorrelate(w,ac,WINDOW,ORDER);
wld(&lpc[1],ac,ORDER); lpc[0]=1.0;
// e=ac[0];
e=0;for(i=0; i<=ORDER; i++) e+=ac[i]*lpc[i]; if (e<0) e=0;
if (e>TOL_OFF) ebuff|=ebit; else ebuff&=~ebit; // update energy bit-buffer
ebit>>=1; if (ebit==0) ebit=ETOPBIT; // circular shift
for (i=0; i<FFT; i++) {real[i]=0; imag[i]=0;}
for (i=0; i<=ORDER; i++) real[i]=lpc[i];
simple_fft(real,imag,realtwiddle,imtwiddle,FFT);
for (i=0; i<FF2; i++) {
b=bin[i];
f2[i]=powf(real[b]*real[b]+imag[b]*imag[b],-0.5);
//f2[i]=powf(f2[i],0.333333);
//f2[i]=powf(f2[i],1.2);
f2[i]=logf(f2[i]);
}
// spectral tilt compensation...
for (i=1; i<FF2; i++) f2[i]=f2[i]*(float)(i+TILT)/TILT;
// fold down to 9 bins...
/*
if (f2[FF2-2]>f2[FF2-3]) f2[FF2-3]=f2[FF2-2]; f2[FF2-2]=0;
if (f2[FF2-4]>f2[FF2-5]) f2[FF2-5]=f2[FF2-4]; f2[FF2-4]=0;
if (f2[FF2-9]>f2[FF2-10]) f2[FF2-10]=f2[FF2-9]; f2[FF2-9]=0;
if (f2[FF2-11]>f2[FF2-12]) f2[FF2-12]=f2[FF2-11]; f2[FF2-11]=0;
if (f2[FF2-13]>f2[FF2-14]) f2[FF2-14]=f2[FF2-13]; f2[FF2-13]=0;
if (f2[FF2-15]>f2[FF2-16]) f2[FF2-16]=f2[FF2-15]; f2[FF2-15]=0;
*/
for (i=0; i<FF2; i++) {
if (f2[i]>6.0) f2[i]=6.0;
f2[i]*=100.0;
}
if (TRACE) { fprintf(stderr,"%.f,",e); for (i=1; i<FF2-1; i++) fprintf(stderr,"%.f,",f2[i]); fprintf(stderr,"\n");}
// calculate frame delta....
delta=0; for (i=1; i<FF2-1; i++) {d=f2[i]-last[i]; delta+=d*d;}
//printf("delta=%f\n",delta);
if (sound==0 && e>TOL_ON && frame>200) { // start recording...
bcopy(last2,&word[wsize],FF2*4); wsize++;
bcopy(last,&word[wsize],FF2*4); wsize++;
sound=1; wsize=0;
bcopy(f2,&word[wsize],FF2*4); wsize++;
bcopy(last,last2,FF2*4);
bcopy(f2,last,FF2*4);
}
else if (sound==1 && e>TOL_OFF) { // continue reading word...
bcopy(f2,&word[wsize],FF2*4); wsize++; if (wsize>200) wsize=200;
bcopy(last,last2,FF2*4);
bcopy(f2,last,FF2*4);
}
else if (sound==1 && ebuff==0) { // finised reading word
// wsize-=8; // remove training silence (2 frame buffer)
if (wsize>4 && wsize<50) {
if (training>0) train();
else closest();
}
sound=0; wsize=0; bcopy(last,last2,FF2*4); bcopy(f2,last,FF2*4);
}
//for (i=1; i<FF2-1; i++) printf("%.0f,",f2[i]); printf(" e=%f\n",e);
laste=e;
frame++; if (frame==37800) exit(0);
}
}
