octave:1> tnewamp1("../build_linux/src/hts1a")
+ 5/ Optionally listen to output
+
+ ~/codec2-dev/build_linux/src$ ./c2sim ../../raw/hts1a.raw --phase0 --postfilter \
+ --amread hts1a_am.out --hmread hts1a_hm.out \
+ --Woread hts1a_Wo.out --hand_voicing hts1a_v.txt -o - \
+ | play -q -t raw -r 8000 -s -2 -
#}
% break into segments of M frames. We have 2 samples spaced M apart
% and interpolate the rest.
- Nfft_phase = 128;
+ Nfft_phase = 128; % note this needs to be 512 (FFT_ENC in codec2 if using --awread)
+ % with --hmread 128 is preferred as less memory/CPU
model_ = zeros(frames, max_amp+2);
voicing_ = zeros(1,frames);
+ Aw = zeros(frames, Nfft_phase);
H = zeros(frames, max_amp);
model_(1,1) = Wo_left = 2*pi/100;
voicing_left = 0;
for k=f-M:f-1
model_(k,:) = resample_rate_L(model_(k,:), interpolated_surface_(k,:), sample_freqs_kHz);
- phase = determine_phase(model_, k, Nfft_phase);
+ Aw(k,:) = determine_phase(model_, k, Nfft_phase);
for m=1:model_(k,2)
b = round(m*model_(k,1)*Nfft_phase/(2*pi)); % map harmonic centre to DFT bin
- H(k,m) = exp(-j*phase(b+1));
+ H(k,m) = exp(j*Aw(k, b+1));
end
end
end
left_vec = right_vec;
end
-
- %model_(1,1:77)
- %model__c(1,1:77)
- %sum(model_(1,1:77)-model__c(1,1:77))
- %[mx mxi] = max(model_(1,1:77)-model__c(1,1:77))
-
- %interpolated_surface_(1,:)
- %interpolated_surface__c(1,:)
- %sum(interpolated_surface_(1,:) - interpolated_surface__c(1,:))
-
-
- %Hm(2,:) - Hm_c(2,:)
- figure(1);
+ figure(1); clf;
mesh(angle(H));
- figure(2);
+ figure(2); clf;
mesh(angle(H_c));
- figure(3);
+ figure(3); clf;
mesh(abs(H - H_c));
check(rate_K_surface, rate_K_surface_c, 'rate_K_surface', 0.01);
check(model_(:,3:max_amp+2), model__c(:,3:max_amp+2), 'rate L Am surface ', 0.1);
check(H, H_c, 'phase surface');
- #{
+ % Save to disk to check synthesis is OK with c2sim
+
+ output_prefix = input_prefix;
+ Am_out_name = sprintf("%s_am.out", output_prefix);
+ fam = fopen(Am_out_name,"wb");
+
+ Wo_out_name = sprintf("%s_Wo.out", output_prefix);
+ fWo = fopen(Wo_out_name,"wb");
+
+ Aw_out_name = sprintf("%s_aw.out", output_prefix);
+ faw = fopen(Aw_out_name,"wb");
+
+ Hm_out_name = sprintf("%s_hm.out", output_prefix);
+ fhm = fopen(Hm_out_name,"wb");
for f=1:frames
printf("%d ", f);
Am_ = zeros(1,max_amp);
Am_(2:L) = Am(1:L-1);
- % optional post filter on linear {Am}, boosts higher amplitudes more than lower,
- % improving shape of formants and reducing muffling. Note energy
- % normalisation
-
- if postfilter
- e1 = sum(Am_(2:L).^2);
- Am_(2:L) = Am_(2:L) .^ 1.5;
- e2 = sum(Am_(2:L).^2);
- Am_(2:L) *= sqrt(e1/e2);
- end
-
fwrite(fam, Am_, "float32");
fwrite(fWo, Wo, "float32");
- if synth_phase
+ % Note we send opposite phase as c2sim expects phase of LPC
+ % analysis filter, just a convention based on historical
+ % development of Codec 2
- % synthesis phase spectra from magnitiude spectra using minimum phase techniques
+ Aw1 = zeros(1, Nfft_phase*2);
+ Aw1(1:2:Nfft_phase*2) = cos(Aw(f,:));
+ Aw1(2:2:Nfft_phase*2) = -sin(Aw(f,:));
+ fwrite(faw, Aw1, "float32");
- fft_enc = 128;
- phase = determine_phase(model_, f, fft_enc);
- assert(length(phase) == fft_enc);
- %Aw = zeros(1, fft_enc*2);
- %Aw(1:2:fft_enc*2) = cos(phase);
- %Aw(2:2:fft_enc*2) = -sin(phase);
+ Hm = zeros(1, 2*max_amp);
+ for m=1:L
+ Hm(2*m+1) = real(H(f,m));
+ Hm(2*m+2) = imag(H(f,m));
+ end
+ fwrite(fhm, Hm, "float32");
+ end
- % sample phase at centre of each harmonic, not 1st entry Hm[1] in octave Hm[0] in C
- % is not used
+ fclose(fam); fclose(fWo); fclose(faw); fclose(fhm);
- Hm = zeros(1, 2*max_amp);
- for m=1:L
- b = round(m*Wo*fft_enc/(2*pi));
- Hm(2*m) = cos(phase(b));
- Hm(2*m+1) = -sin(phase(b));
- end
- fwrite(fhm, Hm, "float32");
- end
+ v_out_name = sprintf("%s_v.txt", output_prefix);
+ fv = fopen(v_out_name,"wt");
+ for f=1:length(voicing__c)
+ fprintf(fv,"%d\n", voicing__c(f));
end
-
- #}
+ fclose(fv);
printf("\n")
endfunction
-% -----------------------------------------------------------------------------------------
-% Linear decimator/interpolator that operates at rate K, includes VQ, post filter, and Wo/E
-% quantisation. Evolved from abys decimator below. Simulates the entire encoder/decoder.
-
-function [model_ voicing_ indexes] = experiment_rate_K_dec(model, voicing)
- max_amp = 80;
- [frames nc] = size(model);
- model_ = zeros(frames, max_amp+3);
- indexes = zeros(frames,4);
-
- M = 4;
-
- % create frames x K surface. TODO make all of this operate frame by
- % frame, or at least M/2=4 frames rather than one big chunk
-
- K = 20;
- [surface sample_freqs_kHz] = resample_const_rate_f_mel(model, K);
- target_surface = surface;
-
- figure(1);
- mesh(surface);
-
- % VQ rate K surface. TODO: If we are decimating by M/2=4 we really
- % only need to do this every 4th frame.
-
- melvq;
- load train_120_vq; m=5;
-
- for f=1:frames
- mean_f(f) = mean(surface(f,:));
- surface_no_mean(f,:) = surface(f,:) - mean_f(f);
- end
- figure(2);
- mesh(surface_no_mean);
-
- [res surface_no_mean_ ind] = mbest(train_120_vq, surface_no_mean, m);
- indexes(:,1:2) = ind;
-
- for f=1:frames
- surface_no_mean_(f,:) = post_filter(surface_no_mean_(f,:), sample_freqs_kHz, 1.5);
- end
- figure(3);
- mesh(surface_no_mean_);
-
- surface_ = zeros(frames, K);
- energy_q = 10 + 40/16*(0:15);
- for f=1:frames
- [mean_f_ indx] = quantise(energy_q, mean_f(f));
- indexes(f,3) = indx - 1;
- %mean_f_ = mean_f(f);
- surface_(f,:) = surface_no_mean_(f,:) + mean_f_;
- end
-
- figure();
- mesh(surface_);
-
- % break into segments of M frames. We have 3 samples in M frame
- % segment spaced M/2 apart and interpolate the rest. This evolved
- % from AbyS scheme below but could be simplified to simple linear
- % interpolation, or using 3 or 4 points but shift of M/2=4 frames.
-
- interpolated_surface_ = zeros(frames, K);
- for f=1:M:frames-M
- left_vec = surface_(f,:);
- right_vec = surface_(f+M,:);
- sample_points = [f f+M];
- resample_points = f:f+M-1;
- for k=1:K
- interpolated_surface_(resample_points,k) = interp_linear(sample_points, [left_vec(k) right_vec(k)], resample_points);
- end
- end
-
- % break into segments for purposes of Wo interpolation
-
- for f=1:M:frames
- % quantise Wo
-
- % UV/V flag is coded using a zero index for Wo, this means we need to
- % adjust Wo index slightly for the lowest Wo V frames
-
- if voicing(f)
- index = encode_log_Wo(model(f,1), 6);
- if index == 0
- index = 1;
- end
- indexes(f,4) = index;
- voicing(f) = 1;
- model_(f,1) = decode_log_Wo(indexes(f,4), 6);
- else
- indexes(f,4) = 0;
- voicing(f) = 0;
- model_(f,1) = 2*pi/100;
- end
- end
-
-
- voicing_ = zeros(1, frames);
- for f=1:M:frames-M
-
- Wo1_ = model_(f,1);
- Wo2_ = model_(f+M,1);
-
- % uncomment to use unquantised values
- %Wo1_ = model(f,1);
- %Wo2_ = model(f+M,1);
-
- if !voicing(f) && !voicing(f+M)
- model_(f:f+M-1,1) = 2*pi/100;
- end
-
- if voicing(f) && !voicing(f+M)
- model_(f:f+M/2-1,1) = Wo1_;
- model_(f+M/2:f+M-1,1) = 2*pi/100;
- voicing_(f:f+M/2-1) = 1;
- end
-
- if !voicing(f) && voicing(f+M)
- model_(f:f+M/2-1,1) = 2*pi/100;
- model_(f+M/2:f+M-1,1) = Wo2_;
- voicing_(f+M/2:f+M-1) = 1;
- end
-
- if voicing(f) && voicing(f+M)
- Wo_samples = [Wo1_ Wo2_];
- model_(f:f+M-1,1) = interp1([f f+M], Wo_samples, f:f+M-1, "linear", 0);
- voicing_(f:f+M-1) = 1;
- end
-
- #{
- printf("f: %d f+M/2: %d Wo: %f %f (%f %%) v: %d %d \n", f, f+M/2, model(f,1), model(f+M/2,1), 100*abs(model(f,1) - model(f+M/2,1))/model(f,1), voicing(f), voicing(f+M/2));
- for i=f:f+M/2-1
- printf(" f: %d v: %d v_: %d Wo: %f Wo_: %f\n", i, voicing(i), voicing_(i), model(i,1), model_(i,1));
- end
- #}
- end
- model_(frames-M:frames,1) = pi/100; % set end frames to something sensible
-
- % enable these to use original (non interpolated) voicing and Wo
- %voicing_ = voicing;
- %model_(:,1) = model(:,1);
-
- model_(:,2) = floor(pi ./ model_(:,1)); % calculate L for each interpolated Wo
- model_ = resample_rate_L(model_, interpolated_surface_, sample_freqs_kHz);
-
-endfunction
void synth_one_frame(codec2_fftr_cfg fftr_inv_cfg, short buf[], MODEL *model, float Sn_[], float Pn[], int prede, float *de_mem, float gain);
void print_help(const struct option *long_options, int num_opts, char* argv[]);
-static void ear_protection(float in_out[], int n);
/*---------------------------------------------------------------------------*\
float lspmelvq_mse = 0.0;
int amread, Woread;
FILE *fam = NULL, *fWo = NULL;
+ int awread;
+ FILE *faw = NULL;
int hmread;
FILE *fhm = NULL;
{ "bpfb", no_argument, &bpfb_en, 1 },
{ "amread", required_argument, &amread, 1 },
{ "hmread", required_argument, &hmread, 1 },
+ { "awread", required_argument, &awread, 1 },
{ "Woread", required_argument, &Woread, 1 },
#ifdef DUMP
{ "dump", required_argument, &dump, 1 },
optarg, strerror(errno));
exit(1);
}
+ } else if(strcmp(long_options[option_index].name, "awread") == 0) {
+ if ((faw = fopen(optarg,"rb")) == NULL) {
+ fprintf(stderr, "Error opening float Aw 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",
codec2_fft(fft_fwd_cfg, a, Aw);
if (hand_voicing) {
- fscanf(fvoicing,"%d\n",&model.voiced);
+ int ret = fscanf(fvoicing,"%d\n",&model.voiced);
+ assert(ret == 1);
}
}
#endif
}
- /* optionally read in Aw FFT vector, we really only care about the phase
- of each entry, used for reading in phases generated by Octave */
-
- if (hmread) {
- int ret = fread(&H, sizeof(float), MAX_AMP, fhm);
- assert(ret == MAX_AMP);
- }
-
- //fprintf(stderr, "frame: %d Wo: %f L: %d v: %d\n", frames, model_dec[i].Wo, model_dec[i].L, model_dec[i].voiced);
if (phase0) {
- sample_phase(&model_dec[i], H, Aw);
+ /* optionally read in Aw, replacing values generated using LPC */
+
+ if (awread) {
+ int ret = fread(Aw, sizeof(COMP), FFT_ENC, faw);
+ assert(ret == FFT_ENC);
+ }
+
+ /* optionally read in Hm directly, bypassing sampling of Aw[] */
+
+ if (hmread) {
+ int ret = fread(H, sizeof(COMP), MAX_AMP, fhm);
+ assert(ret == MAX_AMP);
+ } else {
+ sample_phase(&model_dec[i], H, Aw);
+ }
phase_synth_zero_order(&model_dec[i], ex_phase, H);
}
if (postfilt)
exit(1);
}
-/*---------------------------------------------------------------------------*\
-
- FUNCTION....: ear_protection()
- AUTHOR......: David Rowe
- DATE CREATED: Nov 7 2012
-
- Limits output level to protect ears when there are bit errors or the input
- is overdriven. This doesn't correct or mask bit errors, just reduces the
- worst of their damage.
-
-\*---------------------------------------------------------------------------*/
-
-static void ear_protection(float in_out[], int n) {
- float max_sample, over, gain;
- int i;
-
- /* find maximum sample in frame */
-
- max_sample = 0.0;
- for(i=0; i<n; i++)
- if (in_out[i] > max_sample)
- max_sample = in_out[i];
-
- /* determine how far above set point */
-
- over = max_sample/30000.0;
-
- /* If we are x dB over set point we reduce level by 2x dB, this
- attenuates major excursions in amplitude (likely to be caused
- by bit errors) more than smaller ones */
-
- if (over > 1.0) {
- gain = 1.0/(over*over);
- //fprintf(stderr, "gain: %f\n", gain);
- for(i=0; i<n; i++)
- in_out[i] *= gain;
- }
-}
projects / freetel-svn-tracking.git / commitdiff