diff options
Diffstat (limited to 'octave/tnewamp1.m')
| -rw-r--r-- | octave/tnewamp1.m | 256 |
1 files changed, 256 insertions, 0 deletions
diff --git a/octave/tnewamp1.m b/octave/tnewamp1.m new file mode 100644 index 0000000..6454262 --- /dev/null +++ b/octave/tnewamp1.m @@ -0,0 +1,256 @@ +% tnewamp1.m +% +% Copyright David Rowe 2017 +% This program is distributed under the terms of the GNU General Public License +% Version 2 + +#{ + + Octave script to compare Octave and C versions of newamp1 processing, in order to test C port. + + c2sim -> dump files -> $ ../build_linux/unittest/tnewamp1 -> octave:1> tnewamp1 + Usage: + + 1/ build codec2 with -DDUMP - see codec2-dev/README + + 2/ Generate dump files using c2sim (just need to do this once) + $ cd codec2-dev/build_linux/src + $ ./c2sim ../../raw/hts1a.raw --phase0 --postfilter --dump hts1a --lpc 10 --dump_pitch_e hts1a_pitche.txt + + 3/ Run C version which generates a file of Octave test vectors as output: + + $ cd codec2-dev/build_linux/unittest + $ ./tnewamp1 ../../raw/hts1a.raw + + 4/ Run Octave script to generate Octave test vectors and compare with C. + + 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 - +#} + +function tnewamp1(input_prefix, path_to_unittest="../build_linux/unittest/") + printf("starting tnewamp1.c input_prefix: %s\n", input_prefix); + + visible_flag = 'off'; + newamp_700c; + autotest; + more off; + + max_amp = 80; + postfilter = 0; % optional postfiler that runs on Am, not used atm + synth_phase = 1; + + if nargin == 1 + output_prefix = input_prefix; + end + model_name = strcat(input_prefix,"_model.txt"); + model = load(model_name); + [frames nc] = size(model); + + voicing_name = strcat(input_prefix,"_pitche.txt"); + voicing = zeros(1,frames); + + if exist(voicing_name, "file") == 2 + pitche = load(voicing_name); + voicing = pitche(:, 3); + end + + % Load in C vectors and compare ----------------------------------------- + + load(sprintf("%s/tnewamp1_out.txt", path_to_unittest)); + + K = 20; + [frames tmp] = size(rate_K_surface_c); + [rate_K_surface sample_freqs_kHz] = resample_const_rate_f_mel(model(1:frames,:), K); + + melvq; + load train_120_1.txt; load train_120_2.txt; + train_120_vq(:,:,1)= train_120_1; train_120_vq(:,:,2)= train_120_2; m=5; + m=5; + + eq = zeros(1,K); + for f=1:frames + mean_f(f) = mean(rate_K_surface(f,:)); + rate_K_surface_no_mean(f,:) = rate_K_surface(f,:) - mean_f(f); + [rate_K_vec eq] = front_eq(rate_K_surface_no_mean(f,:), eq); + rate_K_surface_no_mean(f,:) = rate_K_vec; + end + + [res rate_K_surface_no_mean_ ind] = mbest(train_120_vq, rate_K_surface_no_mean, m); + + for f=1:frames + rate_K_surface_no_mean_(f,:) = post_filter(rate_K_surface_no_mean_(f,:), sample_freqs_kHz, 1.5); + end + + rate_K_surface_ = zeros(frames, K); + interpolated_surface_ = zeros(frames, K); + energy_q = create_energy_q; + M = 4; + for f=1:frames + [mean_f_ indx] = quantise(energy_q, mean_f(f)); + indexes(f,3) = indx - 1; + rate_K_surface_(f,:) = rate_K_surface_no_mean_(f,:) + mean_f_; + end + + % simulated decoder + % break into segments of M frames. We have 2 samples spaced M apart + % and interpolate the rest. + + 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; + left_vec = zeros(1,K); + + % decoder runs on every M-th frame, 25Hz frame rate, offset at + % start is to minimise processing delay (thanks Jeroen!) + + for f=M:M:frames + + if voicing(f) + index = encode_log_Wo(model(f,1), 6); + if index == 0 + index = 1; + end + model_(f,1) = decode_log_Wo(index, 6); + else + model_(f,1) = 2*pi/100; + end + + Wo_right = model_(f,1); + voicing_right = voicing(f); + [Wo_ avoicing_] = interp_Wo_v(Wo_left, Wo_right, voicing_left, voicing_right); + + #{ + for i=1:4 + fprintf(stderr, " Wo: %4.3f L: %d v: %d\n", Wo_(i), floor(pi/Wo_(i)), avoicing_(i)); + end + fprintf(stderr," rate_K_vec: "); + for i=1:5 + fprintf(stderr,"%5.3f ", rate_K_surface_(f,i)); + end + fprintf(stderr,"\n"); + #} + + if f > M + model_(f-M:f-1,1) = Wo_; + voicing_(f-M:f-1) = avoicing_; + model_(f-M:f-1,2) = floor(pi ./ model_(f-M:f-1,1)); % calculate L for each interpolated Wo + end + + right_vec = rate_K_surface_(f,:); + + if f > M + sample_points = [f-M f]; + resample_points = f-M:f-1; + for k=1:K + interpolated_surface_(resample_points,k) = interp_linear(sample_points, [left_vec(k) right_vec(k)], resample_points); + end + + for k=f-M:f-1 + model_(k,:) = resample_rate_L(model_(k,:), interpolated_surface_(k,:), sample_freqs_kHz); + 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*Aw(k, b+1)); + end + end + + end + + % update for next time + + Wo_left = Wo_right; + voicing_left = voicing_right; + left_vec = right_vec; + + end + + f = figure(1); clf; + mesh(angle(H)); + f = figure(2); clf; + mesh(angle(H_c(:,1:max_amp))); + f = figure(3); clf; + mesh(abs(H - H_c(:,1:max_amp))); + + passes = 0; tests = 0; + passes += check(eq, eq_c, 'Equaliser', 0.01); tests++; + passes += check(rate_K_surface, rate_K_surface_c, 'rate_K_surface', 0.01); tests++; + passes += check(mean_f, mean_c, 'mean', 0.01); tests++; + passes += check(rate_K_surface_, rate_K_surface__c, 'rate_K_surface_', 0.01); tests++; + passes += check(interpolated_surface_, interpolated_surface__c, 'interpolated_surface_', 0.01); tests++; + passes += check(model_(:,1), model__c(:,1), 'interpolated Wo_', 0.001); tests++; + passes += check(voicing_, voicing__c, 'interpolated voicing'); tests++; + passes += check(model_(:,3:max_amp+2), model__c(:,3:max_amp+2), 'rate L Am surface ', 0.1); tests++; + passes += check(H, H_c(:,1:max_amp), 'phase surface'); tests++; + printf("passes: %d fails: %d\n", passes, tests - passes); + + #{ + % 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"); + + printf("Generating files for c2sim: "); + for f=1:frames + printf(".", f); + Wo = model_(f,1); + L = min([model_(f,2) max_amp-1]); + Am = model_(f,3:(L+2)); + + Am_ = zeros(1,2*max_amp); + Am_(2:L) = Am(1:L-1); + + fwrite(fam, Am_, "float32"); + fwrite(fWo, Wo, "float32"); + + % Note we send opposite phase as c2sim expects phase of LPC + % analysis filter, just a convention based on historical + % development of Codec 2 + + 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"); + + Hm = zeros(1, 2*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 + + fclose(fam); fclose(fWo); fclose(faw); fclose(fhm); + + 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); + #} + +endfunction + + |