streaming working OK, some run time diagnostics

git-svn-id: https://svn.code.sf.net/p/freetel/code@2434 01035d8c-6547-0410-b346-abe4f91aad63

diff --git a/codec2-dev/octave/fsk_horus.m b/codec2-dev/octave/fsk_horus.m
index 910fec0d22b0d93e0f7b75058bfe7fcec0627bfa..2d6ac87dbbd6210db8bd21edba10d343bde30e9d 100644 (file)
--- a/codec2-dev/octave/fsk_horus.m
+++ b/codec2-dev/octave/fsk_horus.m
@@ -39,7 +39,7 @@ function states = fsk_horus_init()
   Rs = states.Rs = 50;
   Ts = states.Ts = Fs/Rs;
   states.nsym = N/Ts;
-  Nmem = states.Nmem  = N+2*Ts;    % one symbol memory in down converted signals to allow for timing adj
+  Nmem = states.Nmem  = N+2*Ts;    % two symbol memory in down converted signals to allow for timing adj
   states.f1_dc = zeros(1,Nmem);
   states.f2_dc = zeros(1,Nmem);
   states.P = 8;                  % oversample rate out of filter
@@ -63,6 +63,8 @@ function states = fsk_horus_init()
   states.uw_thresh = 5*7 - 4; % allow 24bit errors when looking for UW
 
   states.df = 0;
+  states.f1 = 0;
+  states.f2 = 0;
 endfunction
 
 
@@ -155,7 +157,7 @@ function [rx_bits states] = fsk_horus_demod(states, sf)
     %printf("f1: %4.0f Hz f2: %4.0f Hz\n", f1, f2);
   end
 
-  % down convert and filter at 4Rs ------------------------------
+  % down convert and filter at rate P ------------------------------
 
   % update filter (integrator) memory by shifting in nin samples
   
@@ -187,7 +189,7 @@ function [rx_bits states] = fsk_horus_demod(states, sf)
   % integrate over symbol period, which is effectively a LPF, removing
   % the -2Fc frequency image.  Can also be interpreted as an ideal
   % integrate and dump, non-coherent demod.  We run the integrator at
-  % PRs (1/P symbol offsets) to get outputs at a range of different
+  % rate P (1/P symbol offsets) to get outputs at a range of different
   % fine timing offsets.  We calculate integrator output over nsym+1
   % symbols so we have extra samples for the fine timing re-sampler at either
   % end of the array.
@@ -234,7 +236,7 @@ function [rx_bits states] = fsk_horus_demod(states, sf)
   end
   states.nin = next_nin;
 
-  % Re sample integrator outputs using fine timing estimate and linear interpolation
+  % Re-sample integrator outputs using fine timing estimate and linear interpolation
 
   low_sample = floor(rx_timing);
   fract = rx_timing - low_sample;
@@ -467,8 +469,6 @@ endfunction
 function rx_bits_log = demod_file(filename)
   rx = load_raw(filename);
   more off;
-  rand('state',1); 
-  randn('state',1);
   states = fsk_horus_init();
   N = states.N;
   P = states.P;
@@ -574,7 +574,7 @@ endfunction
 %rx_bits = demod_file("~/Desktop/vk5arg-3.wav");
 %rx_bits = demod_file("~/Desktop/fsk_horus_10dB_1000ppm.wav");
 %rx_bits = demod_file("~/Desktop/fsk_horus_6dB_0ppm.wav");
-rx_bits = demod_file("fsk_horus_rx.raw");
+%rx_bits = demod_file("fsk_horus_rx.raw");
 %rx_bits = demod_file("~/Desktop/fsk_horus_20dB_0ppm_20dBfade.wav");
 
 % streaming version: take a buffer of say 1 sec.  demo to bits.  Add to buffer

diff --git a/codec2-dev/octave/fsk_horus_stream.m b/codec2-dev/octave/fsk_horus_stream.m
index 72c9d561f584f0e7bc59379c389b99881a008f11..e081e32703364686576af6573b351e3df27e0a23 100644 (file)
--- a/codec2-dev/octave/fsk_horus_stream.m
+++ b/codec2-dev/octave/fsk_horus_stream.m
@@ -8,7 +8,7 @@
 %
 % usage:
 %  $ chmod 777 fsk_horus_stream.m
-%  $ rec -t raw -r 8000 -s -2 - | ./fsk_horus_stream.m
+%  $ rec -t raw -r 8000 -s -2 -c 1 - -q | ./fsk_horus_stream.m
 
 fsk_horus;  % include library (make sure calls to functions at bottom are commented out)
 
@@ -40,7 +40,8 @@ while c
     nin = states.nin;
     %printf("nin: %d length(rx): %d length(rx_bits_buf): %d \n", nin, length(rx), length(rx_bits_buf));
   endwhile
-  % printf("nin: %d length(rx): %d length(rx_bits_buf): %d \n", nin, length(rx), length(rx_bits_buf));
+  f = (states.f1+states.f2)/2; shift = states.f2 - states.f1;
+  printf("max: %d f: %d shift %d bits: %d\r", max(s), f, shift, length(rx_bits_buf));
 
   % look for complete Horus frame, delimited by 2 unique words