diff options
Diffstat (limited to 'src/newamp2.c')
| -rw-r--r-- | src/newamp2.c | 735 |
1 files changed, 351 insertions, 384 deletions
diff --git a/src/newamp2.c b/src/newamp2.c index bba4cc9..0b5f032 100644 --- a/src/newamp2.c +++ b/src/newamp2.c @@ -2,8 +2,8 @@ FILE........: newamp2.c AUTHOR......: Thomas Kurin and Stefan Erhardt - INSTITUTE...: Institute for Electronics Engineering, University of Erlangen-Nuremberg - DATE CREATED: July 2018 + 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" @@ -30,540 +30,507 @@ */ +#include "newamp2.h" + #include <assert.h> +#include <math.h> #include <stdio.h> #include <stdlib.h> #include <string.h> -#include <math.h> #include "defines.h" -#include "phase.h" -#include "quantise.h" #include "mbest.h" #include "newamp1.h" -#include "newamp2.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 + 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. + 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]; +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 + 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; +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 */ + /* 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]); + 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]; } - - /* clip between peak and peak -50dB, to reduce dynamic range */ + 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]); + } - for(m=1; m<=model->L; m++) { - if (AmdB[m] < (AmdB_peak-50.0)) { - AmdB[m] = AmdB_peak-50.0; - } + /* 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); + 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 + 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) -{ +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]; + int index[1]; /* codebook is compiled for a fixed K */ - //assert(ndim == newamp2vq_cb[0].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; + 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); + 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; - + // 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 + 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]); - } +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 + 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]; - } +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 + 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; +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; } - else { - indexes[3] = 0; + if (index == 63) { + index = 62; } - if(plosive != 0){ - indexes[3] = 63; - } - } - + 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 + 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]; - } +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]; - post_filter_newamp1(rate_K_vec_no_mean_, rate_K_sample_freqs_kHz, K, pf_gain); + for (k = 0; k < K; k++) { + rate_K_vec_no_mean_[k] = codebook1[(NEWAMP2_16K_K + 1) * n1 + k]; + } - *mean_ = newamp2_energy_cb[0].cb[indexes[2]]; + post_filter_newamp1(rate_K_vec_no_mean_, rate_K_sample_freqs_kHz, K, pf_gain); - for(k=0; k<K; k++) { - rate_K_vec_[k] = rate_K_vec_no_mean_[k] + *mean_; - } + *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 + 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) + 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; - } +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 + 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]; - } - } - } - - } -} +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 + 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[], +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; + 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); + } } - //indexes[3]=63 (= Plosive) and unvoiced + // No Plosive, standard resample 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); + 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; + /* 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; } - |