% In general, this function processes a bunch of amplitudes, we then
% use c2sim to hear the results. Bunch of different experiments below
-function surface = newamp1_batch(input_prefix, output_prefix, mode)
+function surface = newamp1_batch(input_prefix, varargin)
newamp;
more off;
max_amp = 160;
- postfilter = 0; % optional postfiler that runs on Am, not used atm
+
+ % defaults
+
synth_phase = 1;
+ output_prefix = input_prefix;
+ vq_filename = "";
+ mean_removal = 0;
+ mode = "const";
+
+ % parse variable argument list
- if nargin == 1
- output_prefix = input_prefix;
+ if (length (varargin) > 0)
+
+ % check for the "output_prefix" option
+
+ ind = arg_exists(varargin, "output_prefix");
+ if ind
+ output_prefix = varargin{ind+1};
+ end
+ ind = arg_exists(varargin, "mode");
+ if ind
+ mode = varargin{ind+1};
+ end
+ ind = arg_exists(varargin, "nomean");
+ if ind
+ mean_removal = 1;
+ end
+ ind = arg_exists(varargin, "vq");
+ if ind
+ vq_filename = varargin{ind+1};
+ end
end
+
model_name = strcat(input_prefix,"_model.txt");
model = load(model_name);
[frames nc] = size(model);
if strcmp(mode, 'mel')
[model_ surface] = experiment_mel_freq(model, 0, 1, voicing);
end
+ if strcmp(mode, 'const')
+ [model_ surface] = experiment_const_freq(model, mean_removal, vq_filename);
+ end
+ if strcmp(mode, 'piecewise')
+ model_ = experiment_piecewise(model);
+ end
+ if strcmp(mode, 'dct')
+ model_ = experiment_dct(model);
+ end
% ----------------------------------------------------
for f=1:frames
%printf("%d ", f);
- Wo = model_(f,1);
- L = min([model_(f,2) max_amp-1]);
- Am = model_(f,3:(L+2));
-
- 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
+ Wo = model_(f,1); L = min([model_(f,2) max_amp-1]); Am = model_(f,3:(L+2));
- 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");
+ Am_ = zeros(1,max_amp); Am_(2:L) = Am(1:L-1); fwrite(fam, Am_, "float32");
fwrite(fWo, Wo, "float32");
if synth_phase
fft_enc = 256;
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);
% sample phase at centre of each harmonic, not 1st entry Hm[1] in octave Hm[0] in C
% is not used
endfunction
+function ind = arg_exists(v, str)
+ ind = 0;
+ for i=1:length(v)
+ if strcmp(v{i}, str)
+ ind = i;
+ end
+ end
+endfunction
+
+% Basic unquantised rate K linear sampling then back to rate L. Used for generating
+% training vectors and testing vector quntisers.
+
+function [model_ rate_K_surface] = experiment_const_freq(model, mean_removal, vq_filename)
+ [frames nc] = size(model);
+ Fs = 8000;
+
+ melvq;
+ if length(vq_filename)
+ x = load(vq_filename); m=5;
+ vq = x.vq;
+ size(vq)
+ end
+
+ rate_K_sample_freqs_kHz = [0.1:0.1:3.9];
+ K = length(rate_K_sample_freqs_kHz);
+ rate_K_surface = resample_const_rate_f(model, rate_K_sample_freqs_kHz, Fs);
+
+ mean_f = zeros(1,frames);
+ if mean_removal
+ for f=1:frames
+ mean_f(f) = mean(rate_K_surface(f,:));
+ rate_K_surface(f,:) -= mean_f(f);
+ end
+ end
+
+ if length(vq_filename)
+ [res rate_K_surface_ ind] = mbest(vq, rate_K_surface, m);
+ sd_per_frame = std(res');
+ energy = zeros(1,frames);
+ for f=1:frames
+ L = model(f,2);
+ energy(f) = 10*log10(sum( model(f,3:(L+2)) .^ 2 ));
+ end
+ figure(1); subplot(211); plot(energy); subplot(212); plot(sd_per_frame);
+ figure(2); hist(sd_per_frame);
+ printf("VQ rms SD: %3.2f\n", mean(sd_per_frame));
+ else
+ rate_K_surface_ = rate_K_surface;
+ end
+
+ if mean_removal
+ for f=1:frames
+ rate_K_surface_(f,:) += mean_f(f);
+ end
+ end
+
+ [model_ AmdB_] = resample_rate_L(model, rate_K_surface_, rate_K_sample_freqs_kHz, Fs);
+
+ sum_sd = 0;
+ for f=1:frames
+ L = model(f,2);
+ AmdB = 20*log10(model(f,3:(L+2)));
+ sum_sd = std(AmdB(1:L) - AmdB_(f,1:L));
+ end
+ printf("rate K resampling SD: %3.2f\n", mean(sum_sd));
+
+endfunction
+
+
+function model_ = experiment_dct(model)
+
+ [frames tmp] = size(model); max_amp = 160;
+
+ for f=1:frames
+ printf("%d ", f);
+ Wo = model(f,1);
+ L = min([model(f,2) max_amp-1]);
+ Am = model(f,3:(L+2));
+ AmdB = 20*log10(Am);
+
+ % fit model
+
+ D = dct(AmdB);
+ E = zeros(1,L);
+ E(1:min(20,L)) = D(1:min(20,L));
+ AmdB_ = idct(D);
+
+ model_(f,1) = Wo; model_(f,2) = L; model_(f,3:(L+2)) = 10 .^ (AmdB_(1:L)/20);
+ end
+
+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.
endfunction
+
% mel spaced sampling, differential in time VQ. Curiously, couldn't
% get very good results out of this, I suspect a bug
quantise.
todo: get this working again
+#}
function model_ = experiment_piecewise(model)
- % encoder loop ------------------------------------------------------
+
+ [frames tmp] = size(model); max_amp = 160;
fvec_log = []; amps_log = [];
fvec_log = [fvec_log; fvec];
amps_log = [amps_log; amps];
- model_(f,1) = Wo; model_(f,2) = L; model_(f,3:(L+2)) = 10 .^ (AmdB(1:L)/20);
+ model_(f,1) = Wo; model_(f,2) = L; model_(f,3:(L+2)) = 10 .^ (AmdB_(1:L)/20);
end
- for f=1:frames
- if 0
- if (f > 1) && (e(f) > (e(f-1)+3))
- decimate = 2;
- else
- decimate = 4;
- end
- end
- printf("%d ", decimate);
-
- AmdB_ = decimate_frame_rate(model_, decimate, f, frames);
- L = length(AmdB_);
-
- Am_ = zeros(1,max_amp);
- Am_(2:L) = 10 .^ (AmdB_(1:L-1)/20); % C array doesnt use A[0]
- fwrite(fam, Am_, "float32");
- fwrite(fWo, Wo, "float32");
- end
endfunction
-#}
-
% early test, devised to test rate K<->L changes along frequency axis
projects / freetel-svn-tracking.git / commitdiff