refactored model_from_indexes and homebrew linear interpolator to match C code

git-svn-id: https://svn.code.sf.net/p/freetel/code@2947 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/codec2-dev/octave/newamp.m b/codec2-dev/octave/newamp.m
index 3a1512038821f14c9479900ca80f6ec364ac7300..7fe5b43747ef505672dd5c1f788c78a1e1725f16 100644 (file)
--- a/codec2-dev/octave/newamp.m
+++ b/codec2-dev/octave/newamp.m
@@ -45,6 +45,21 @@ function y = interp_para(xp, yp, x)
 endfunction
 
 
+% simple linear interpolator
+
+function y = interp_linear(xp, yp, x)
+  assert( (length(xp) == 2) && (length(yp) == 2) );
+
+  m = (yp(2) - yp(1))/(xp(2) - xp(1));
+  c = yp(1) - m*xp(1);
+
+  y = zeros(1,length(x));
+  for i=1:length(x)
+    y(i) = m*x(i) + c;
+  end
+endfunction
+
+
 % quantise input sample to nearest value in table, optionally return binary code
 
 function [quant_out best_i bits] = quantise(levels, quant_in)
@@ -216,6 +231,7 @@ function [model_ AmdB_] = resample_rate_L(model, rate_K_surface, rate_K_sample_f
   for f=1:frames
     Wo = model(f,1);
     L = model(f,2);
+    f
     rate_L_sample_freqs_kHz = (1:L)*Wo*4/pi;
     
     % back down to rate L

diff --git a/codec2-dev/octave/newamp1_batch.m b/codec2-dev/octave/newamp1_batch.m
index d72d62b6ee10992be0f6d6eb94fd9b68f89db292..0280feaafd66eb70d45eb35e65f04fff580c0a25 100644 (file)
--- a/codec2-dev/octave/newamp1_batch.m
+++ b/codec2-dev/octave/newamp1_batch.m
@@ -312,50 +312,45 @@ endfunction
 % decoder and an important step in proving everything works
  
 function [model_ voicing_] = model_from_indexes(indexes)
-  max_amp = 80;
+  max_amp = 80;  K = 20;  M = 4;
+
   [frames nc] = size(indexes);
   model = model_ = zeros(frames, max_amp+3);
-  K = 20;
   sample_freqs_kHz = mel_sample_freqs_kHz(K);
-  M = 8;
   energy_q = 10 + 40/16*(0:15);
 
   melvq;
   load train_120_vq;
 
+  % decode vector quantised surface
+
   surface_no_mean_ = zeros(frames,K);
   surface_ = zeros(frames, K);
-  for f=1:M/2:frames
+  for f=1:M:frames
     surface_no_mean_(f,:) = train_120_vq(indexes(f,1),:,1) + train_120_vq(indexes(f,2),:,2);
     surface_no_mean_(f,:) = post_filter(surface_no_mean_(f,:), sample_freqs_kHz, 1.5);
     mean_f_ = energy_q(indexes(f,3)+1);
     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.
+  % break into segments of M frames.  We have 2 samples spaced M apart
+  % and interpolate the rest.
   
   interpolated_surface_ = zeros(frames, K);
-  for f=1:M/2:frames-M
+  for f=1:M:frames-M
     left_vec = surface_(f,:);
-    m = f+M/2;
-    centre_vec = surface_(m,:);
     right_vec = surface_(f+M,:);
-    %sample_points = [f m f+M];
-    sample_points = [f m];
+    sample_points = [f f+M];
     resample_points = f:f+M-1;
     for k=1:K
-      % interpolated_surface_(resample_points,k) = interp1(sample_points, [left_vec(k) centre_vec(k) right_vec(k)], resample_points, "spline", 0);
-      interpolated_surface_(resample_points,k) = interp1(sample_points, [left_vec(k) centre_vec(k)], resample_points, "spline", 0);
+      interpolated_surface_(resample_points,k) = interp_linear(sample_points, [left_vec(k) right_vec(k)], resample_points);
     end    
   end
 
   % recover Wo and voicing
 
   voicing = zeros(1, frames);
-  for f=1:M/2:frames-M/2
+  for f=1:M:frames-M
     if indexes(f,4) == 0
       voicing(f) = 0;
       model(f,1) = 2*pi/100;
@@ -365,34 +360,34 @@ function [model_ voicing_] = model_from_indexes(indexes)
     end
   end
 
-  % break into M/2 segments for purposes of Wo interpolation
+  % break into M segments for purposes of Wo interpolation
 
   voicing_ = zeros(1, frames);
-  for f=1:M/2:frames-M/2
+  for f=1:M:frames-M
 
     Wo1_ = model(f,1);
-    Wo2_ = model(f+M/2,1);
+    Wo2_ = model(f+M,1);
 
-    if !voicing(f) && !voicing(f+M/2)
-       model_(f:f+M/2-1,1) = 2*pi/100;
+    if !voicing(f) && !voicing(f+M)
+       model_(f:f+M-1,1) = 2*pi/100;
     end
 
-    if voicing(f) && !voicing(f+M/2)
-       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;
+    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/2)
-       model_(f:f+M/4-1,1) = 2*pi/100;
-       model_(f+M/4:f+M/2-1,1) = Wo2_;
-       voicing_(f+M/4:f+M/2-1) = 1;
+    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/2)
+    if voicing(f) && voicing(f+M)
       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;
+      model_(f:f+M-1,1) = interp_linear([f f+M], Wo_samples, f:f+M-1);
+      voicing_(f:f+M-1) = 1;
     end
 
     #{
@@ -402,11 +397,11 @@ function [model_ voicing_] = model_from_indexes(indexes)
     end
     #}
   end
-  model_(frames-M/2:frames,1) = pi/100; % set end frames to something sensible
+  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_(:,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);