quantised Wo to 6 bits and E to 4 bits, sounds OK

git-svn-id: https://svn.code.sf.net/p/freetel/code@2938 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/codec2-dev/octave/newamp1_batch.m b/codec2-dev/octave/newamp1_batch.m
index 04c3d3342d4b29b5885b8dbafb6cb92359e77540..81d0b03366b980df7b5d274009fa10975ed65613 100644 (file)
--- a/codec2-dev/octave/newamp1_batch.m
+++ b/codec2-dev/octave/newamp1_batch.m
@@ -7,13 +7,24 @@
 % Octave script to batch process model parameters using the new
 % amplitude model.  Used for generating samples we can listen to.
 %
-% Usage:
-%   ~/codec2-dev/build_linux/src$ ./c2sim ../../raw/hts1a.raw --dump hts1a
-%   $ cd ~/codec2-dev/octave
-%   octave:14> newamp1_batch("../build_linux/src/hts1a")
-%   ~/codec2-dev/build_linux/src$ ./c2sim ../../raw/hts1a.raw --amread hts1a_am.out -o - | play -t raw -r 8000 -s -2 -
-% Or with a little more processing:
-%   codec2-dev/build_linux/src$ ./c2sim ../../raw/hts2a.raw --amread hts2a_am.out --awread hts2a_aw.out --phase0 --postfilter --Woread hts2a_Wo.out -o - | play -q -t raw -r 8000 -s -2 -
+
+#{
+
+  Usage:
+
+    ~/codec2-dev/build_linux/src$ ./c2sim ../../raw/hts1a.raw --dump hts1a
+    $ cd ~/codec2-dev/octave
+    octave:14> newamp1_batch("../build_linux/src/hts1a")
+    ~/codec2-dev/build_linux/src$ ./c2sim ../../raw/hts1a.raw --amread hts1a_am.out -o - | play -t raw -r 8000 -s -2 -
+ 
+  Or with a little more processing, first dump energy and voicing, the import Wo, voicing, phase spectra:
+
+    $ ./c2sim ../../raw/vk5qi.raw --phase0 --postfilter --dump vk5qi --lpc 10 --dump_pitch_e vk5qi_pitche.txt
+    octave:14> newamp1_batch("../build_linux/src/vk5qi", "../build_linux/src/vk5qi_am_Wo.out");
+    $ ./c2sim ../../raw/vk5qi.raw --phase0 --postfilter --amread vk5qi_am_Wo.out --awread vk5qi_aw.out --Woread vk5qi_Wo.out --hand_voicing vk5qi_v.txt -o - | play -q -t raw -r 8000 -s -2 -
+
+#}
+
 
 % process a whole file and write results
 % TODO: 
@@ -55,7 +66,8 @@ function surface = newamp1_batch(samname, optional_Am_out_name, optional_Aw_out_
 
   %[model_ surface] = experiment_mel_freq(model, 1, 1, voicing);
   %model_ = experiment_dec_abys(model, 8, 1, 1, 1, voicing);
-  [model_ voicing_] = experiment_rate_K_dec(model, voicing);
+  [model_ voicing_ indexes] = experiment_rate_K_dec(model, voicing);
+  indexes
 
   %model_ = experiment_dec_linear(model_);
   %model_ = experiment_energy_rate_linear(model, 1, 0);
@@ -328,11 +340,12 @@ endfunction
 % Linear decimator/interpolator that operates at rate K, includes VQ, post filter, and Wo/E
 % quantisation.  Evevoled from abys decimator below.
  
-function [model_ voicing_ ] = experiment_rate_K_dec(model, voicing)
+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 = 8;
  
   % create frames x K surface.  TODO make all of this operate frame by
@@ -354,21 +367,26 @@ function [model_ voicing_ ] = experiment_rate_K_dec(model, voicing)
   end
 
   [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
     
   surface_ = zeros(frames, K);
-  for f=1:frames
-    surface_(f,:) = surface_no_mean_(f,:) + mean_f(f);
+  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
 
   % 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,:);
@@ -387,39 +405,53 @@ function [model_ voicing_ ] = experiment_rate_K_dec(model, voicing)
   voicing_ = zeros(1, frames);
   for f=1:M/2:frames-M/2
 
+    % quantise Wo
+
+    index = encode_log_Wo(model(f,1), 6);
+    Wo1_ = decode_log_Wo(index, 6);
+    indexes(f,4) = index;
+    index = encode_log_Wo(model(f+M/2,1), 6);
+    Wo2_ = decode_log_Wo(index, 6);
+
+    % uncomment to use unquantised values
+    %Wo1_ = model(f,1);
+    %Wo2_ = model(f+M/2,1);
+
     if !voicing(f) && !voicing(f+M/2)
        model_(f:f+M/2-1,1) = 2*pi/100;
     end
 
     if voicing(f) && !voicing(f+M/2)
-       model_(f:f+M/4-1,1) = model(f,1);
+       model_(f:f+M/4-1,1) = Wo1_;
        model_(f+M/4:f+M/2-1,1) = 2*pi/100;
        voicing_(f:f+M/4-1) = 1;
     end
 
     if !voicing(f) && voicing(f+M/2)
        model_(f:f+M/4-1,1) = 2*pi/100;
-       model_(f+M/4:f+M/2-1,1) = model(f+M/2,1);
+       model_(f+M/4:f+M/2-1,1) = Wo2_;
        voicing_(f+M/4:f+M/2-1) = 1;
     end
 
     if voicing(f) && voicing(f+M/2)
-      Wo_samples = [model(f,1) model(f+M/2,1)];
+      Wo_samples = [Wo1_ Wo2_];
       model_(f:f+M/2-1,1) = interp1([f f+M/2], Wo_samples, f:f+M/2-1, "linear", 0);
       voicing_(f:f+M/2-1) = 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));
+    #{
+    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/2:frames,1) = pi/100; % set end frames to something sensible
 
-  voicing_ = voicing;
-  model_(:,1) = model(:,1);
-  %model_(221:225,1) = model(221:225,1);
-  %model_(223:224,1) = model(223:224,1);
+  % 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);
 

diff --git a/codec2-dev/src/c2sim.c b/codec2-dev/src/c2sim.c
index 9a15be9bbf9cc3f844dc0d59f54f51b53b79c5f4..e9d194e379fdf817ad4ea8a4a96616277ba6a2db 100644 (file)
--- a/codec2-dev/src/c2sim.c
+++ b/codec2-dev/src/c2sim.c
@@ -839,7 +839,7 @@ int main(int argc, char *argv[])
                     assert(ret == FFT_ENC);
                 }
 
-                fprintf(stderr, "frame: %d Wo: %f L: %d v: %d\n", frames, model_dec[i].Wo, model_dec[i].L,  model_dec[i].voiced);
+                //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)
                     phase_synth_zero_order(fft_fwd_cfg, &model_dec[i], ex_phase, Aw);
                 if (postfilt)