From: drowe67 <drowe67@01035d8c-6547-0410-b346-abe4f91aad63>
Date: Tue, 1 Jul 2014 04:08:12 +0000 (+0000)
Subject: removing a bunch of old unused directories
X-Git-Url: http://git.whiteaudio.com/gitweb/?a=commitdiff_plain;h=c35f16f450fd56b821922bfee17cf5fcc8d5a039;p=freetel-svn-tracking.git

removing a bunch of old unused directories

git-svn-id: https://svn.code.sf.net/p/freetel/code@1720 01035d8c-6547-0410-b346-abe4f91aad63
---

diff --git a/codec2-dev/CMakeLists.txt b/codec2-dev/CMakeLists.txt
index ae10d9c2..c98bb330 100644
--- a/codec2-dev/CMakeLists.txt
+++ b/codec2-dev/CMakeLists.txt
@@ -84,7 +84,7 @@ set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -fPIC -Wall")
 #a
 option(BUILD_SHARED_LIBS
     "Build shared library. Set to OFF for static library." ON)
-option(UNITTEST "Build unittest binaries." OFF)
+option(UNITTEST "Build unittest binaries." ON)
 option(INSTALL_EXAMPLES "Install example code." OFF)
 if(INSTALL_EXAMPLES)
     install(DIRECTORY octave raw script voicing wav
diff --git a/codec2-dev/README_fdmdv.txt b/codec2-dev/README_fdmdv.txt
index 81c8c4df..110703a0 100644
--- a/codec2-dev/README_fdmdv.txt
+++ b/codec2-dev/README_fdmdv.txt
@@ -33,7 +33,7 @@ $ cd src
  
 3. Send 14000 modulated bits (10 seconds) to the demod and count errors:
 
-    $  ./fdmdv_get_test_bits - 14000 | ./fdmdv_mod - - | ./fdmdv_demod - - demod_dump.txt | ./fdmdv_put_test_bits -
+    $  ./fdmdv_get_test_bits - 14000 | ./fdmdv_mod - - | ./fdmdv_demod - - 14 demod_dump.txt | ./fdmdv_put_test_bits -
 
     Use Octave to look at plots of 1 second (1400 bits) of modem operation:
 
diff --git a/codec2-dev/fltk/Makefile b/codec2-dev/fltk/Makefile
deleted file mode 100644
index f29d2cd6..00000000
--- a/codec2-dev/fltk/Makefile
+++ /dev/null
@@ -1,19 +0,0 @@
-# Requires FLTK 1.3 & Portaudio V19
-
-FLTK_VER = $(shell fltk-config --api-version)
-ifneq ($(FLTK_VER),1.3)
-$(error Must use FLTK version 1.3, you have $(FLTK_VER))
-endif
-
-FLTK_CFLAGS += $(shell fltk-config --ldstaticflags)
-CFLAGS = -O3 -g -Wall
-LIBS = ../src/.libs/libcodec2.a -lm -lrt -lportaudio -pthread
-LC2POC_C = fl_fdmdv.cxx
-
-all: fl_fdmdv
-
-fl_fdmdv: Makefile $(LC2POC_C)
-	g++ $(LC2POC_C) -I../src/ -o fl_fdmdv $(CFLAGS) $(FLTK_CFLAGS) $(LIBS)
-
-clean:
-	rm -f fl_fdmdv
diff --git a/codec2-dev/fltk/fl_fdmdv.cxx b/codec2-dev/fltk/fl_fdmdv.cxx
deleted file mode 100644
index 30c67aab..00000000
--- a/codec2-dev/fltk/fl_fdmdv.cxx
+++ /dev/null
@@ -1,1158 +0,0 @@
-/*
-  fl_fdmdv.cxx
-  Created 14 June 2012
-  David Rowe
-
-  Fltk 1.3 based GUI program to prototype FDMDV & Codec 2 integration
-  issues such as:
-
-    + spectrum, waterfall, and other FDMDV GUI displays
-    + integration with real time audio I/O using portaudio
-    + what we do with audio when out of sync
-*/
-
-#include <assert.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <unistd.h>
-
-#include <FL/Fl.H>
-#include <FL/Fl_Window.H>
-#include <FL/Fl_Box.H>
-#include <FL/fl_draw.H>
-#include <FL/Fl_Group.H>
-#include <FL/names.h>
-
-#include "portaudio.h"
-
-#include "fdmdv.h"
-#include "codec2.h"
-
-#define MIN_DB             -40.0 
-#define MAX_DB               0.0
-#define BETA                 0.1  // constant for time averageing spectrum data
-#define MIN_HZ               0
-#define MAX_HZ            4000
-#define WATERFALL_SECS_Y     5    // number of seconds respresented by y axis of waterfall
-#define DT                   0.02 // time between samples 
-#define FS                8000    // FDMDV modem sample rate
-
-#define SCATTER_MEM       (FDMDV_NSYM)*50
-#define SCATTER_X_MAX        3.0
-#define SCATTER_Y_MAX        3.0
-
-// main window params
-
-#define W                  1200
-#define W3                 (W/3)
-#define H                  600
-#define H2                 (H/2)
-#define SP                  20
-
-// sound card
-
-#define SAMPLE_RATE  48000                        /* 48 kHz sampling rate rec. as we
-				                     can trust accuracy of sound
-				                     card                                    */
-#define N8           FDMDV_NOM_SAMPLES_PER_FRAME  /* processing buffer size at 8 kHz         */
-#define MEM8 (FDMDV_OS_TAPS/FDMDV_OS)
-#define N48          (N8*FDMDV_OS)                /* processing buffer size at 48 kHz        */
-#define NUM_CHANNELS 2                            /* I think most sound cards prefer stereo,
-				                     we will convert to mono                 */
-
-#define BITS_PER_CODEC_FRAME (2*FDMDV_BITS_PER_FRAME)
-#define BYTES_PER_CODEC_FRAME (BITS_PER_CODEC_FRAME/8)
-
-// forward class declarations
-
-class Spectrum;
-class Waterfall;
-class Scatter;
-class Scalar;
-
-// Globals --------------------------------------
-
-char         *fin_name = NULL;
-char         *fout_name = NULL;
-char         *sound_dev_name = NULL;
-FILE         *fin = NULL;
-FILE         *fout = NULL;
-struct FDMDV *fdmdv;
-struct CODEC2 *codec2;
-float         av_mag[FDMDV_NSPEC]; // shared between a few classes
-
-// GUI variables --------------------------------
-
-Fl_Group     *agroup;
-Fl_Window    *window;
-Fl_Window    *zoomSpectrumWindow = NULL;
-Fl_Window    *zoomWaterfallWindow = NULL;
-Spectrum     *aSpectrum;
-Spectrum     *aZoomedSpectrum;
-Waterfall    *aWaterfall;
-Waterfall    *aZoomedWaterfall;
-Scatter      *aScatter;
-Scalar       *aTimingEst;
-Scalar       *aFreqEst;
-Scalar       *aSNR;
-int          zoom_spectrum = 0;
-
-// Main processing loop states ------------------
-
-float  Ts = 0.0;
-short  input_buf[2*FDMDV_NOM_SAMPLES_PER_FRAME];
-int    n_input_buf = 0;
-int    nin = FDMDV_NOM_SAMPLES_PER_FRAME;
-short *output_buf;
-int    n_output_buf = 0;
-int    codec_bits[2*FDMDV_BITS_PER_FRAME];
-int    state = 0;
-
-// Portaudio states -----------------------------
-
-PaStream *stream = NULL;
-PaError err;
-
-typedef struct {
-    float               in48k[FDMDV_OS_TAPS + N48];
-    float               in8k[MEM8 + N8];
-} paCallBackData;
-
-// Class for each window type  ------------------
-
-class Spectrum: public Fl_Box {
-protected:
-    int handle(int event) {
-
-	//  detect a left mouse down if inside the spectrum window
-
-	if ((event == FL_NO_EVENT) && (Fl::event_button() == 1)) {
-	    if ((Fl::event_x() > x()) && (Fl::event_x() < (x() + w())) &&
-		(Fl::event_y() > y()) && (Fl::event_y() < (y() + h()))) {
-
-		// show zoomed spectrum window
-
-		zoomSpectrumWindow->show();
-	    }
-	    
-	}
-	return 0;
-    }
-
-    void draw() {
-	float x_px_per_point = 0.0;
-	float y_px_per_dB = 0.0;
-	int   i, x1, y1, x2, y2;
-	float mag1, mag2;
-	char  label[20];
-	float px_per_hz;
-
-	Fl_Box::draw();
-	fl_color(FL_BLACK);
-	fl_rectf(x(),y(),w(),h());
-	fl_color(FL_GREEN);
-	fl_line_style(FL_SOLID);
-
-	fl_push_clip(x(),y(),w(),h());
-	//printf("%d %d\n", w(), h());
-	x_px_per_point = (float)w()/FDMDV_NSPEC;
-	y_px_per_dB = (float)h()/(MAX_DB - MIN_DB);
-
-	// plot spectrum
-
-	for(i=0; i<FDMDV_NSPEC-1; i++) {
-	    mag1 = av_mag[i];
-	    mag2 = av_mag[i+1];
-
-	    x1 = x() + i*x_px_per_point;
-	    y1 = y() + -mag1*y_px_per_dB;
-	    x2 = x() + (i+1)*x_px_per_point;
-	    y2 = y() + -mag2*y_px_per_dB;
-	    fl_line(x1,y1,x2,y2);   
-	}
-
-	// y axis graticule
-
-	fl_line_style(FL_DOT);
-	for(i=MIN_DB; i<MAX_DB; i+=10) {
-	    x1 = x();
-	    y1 = y() + -i*y_px_per_dB;
-	    x2 = x() + w();
-	    y2 = y1;
-	    //printf("%d %d %d %d\n", x1, y1, x2, y2);
-	    fl_line(x1,y1,x2,y2);   
-	    sprintf(label, "%d", i);
-	    fl_draw(label, x1, y1);
-	}
-
-	// x axis graticule
-
-	px_per_hz = (float)w()/(MAX_HZ-MIN_HZ);
-	fl_line_style(FL_DOT);
-	for(i=500; i<MAX_HZ; i+=500) {
-	    x1 = x() + i*px_per_hz;
-	    y1 = y();
-	    x2 = x1;
-	    y2 = y() + h();
-	    //printf("i=%d %d %d %d %d\n", i, x1, y1, x2, y2);
-	    fl_line(x1,y1,x2,y2);   
-	    sprintf(label, "%d", i);
-	    fl_draw(label, x1, y2);
-	}
-
-	fl_pop_clip();
-    }
-
-public:
-    Spectrum(int x, int y, int w, int h): Fl_Box(x, y, w, h, "Spectrum")
-    {
-	align(FL_ALIGN_TOP);
-	labelsize(10);
-    };
-
-};
-
-
-/*
-
-  Notes:
-
-  The height h() pixels represents WATERFALL_SECS_Y of data.  Every DT
-  seconds we get a vector of FDMDV_NSPEC spectrum samples which we use
-  to update the last row.  The height of each row is dy pixels, which
-  maps to DT seconds.  We call each dy high rectangle of pixels a
-  block.
-
-*/
-
-class Waterfall: public Fl_Box {
-protected:
-
-    int       prev_w, prev_h;
-    unsigned *pixel_buf;
-    unsigned  heatmap_lut[256];
-    int       greyscale;
-
-    void new_pixel_buf(int w, int h) {
-	int buf_sz, i;
-
-	prev_w = w; prev_h = h;
-	buf_sz = h*w;
-	pixel_buf = new unsigned[buf_sz];
-	for(i=0; i<buf_sz; i++)
-	    pixel_buf[i] = 0;
-    }
-    
-    int handle(int event) {
-
-	//  detect a left mouse down if inside the window
-
-	if ((event == FL_NO_EVENT) && (Fl::event_button() == 1)) {
-	    if ((Fl::event_x() > x()) && (Fl::event_x() < (x() + w())) &&
-		(Fl::event_y() > y()) && (Fl::event_y() < (y() + h()))) {
-
-		// show zoomed spectrum window
-
-		zoomWaterfallWindow->show();
-	    }
-	    
-	}
-	return 0;
-    }
-
-    // map val to a rgb colour
-    // from http://eddiema.ca/2011/01/21/c-sharp-heatmaps/
-
-    unsigned heatmap(float val, float min, float max) {
-	unsigned r = 0;
-	unsigned g = 0;
-	unsigned b = 0;
-
-	val = (val - min) / (max - min);
-	if(val <= 0.2) {
-	    b = (unsigned)((val / 0.2) * 255);
-	} else if(val >  0.2 &&  val <= 0.7) {
-	    b = (unsigned)((1.0 - ((val - 0.2) / 0.5)) * 255);
-	}
-	if(val >= 0.2 &&  val <= 0.6) {
-	    g = (unsigned)(((val - 0.2) / 0.4) * 255);
-	} else if(val >  0.6 &&  val <= 0.9) {
-	    g = (unsigned)((1.0 - ((val - 0.6) / 0.3)) * 255);
-	}
-	if(val >= 0.5) {
-	    r = (unsigned)(((val - 0.5) / 0.5) * 255);
-	}
-    
-	//printf("%f %x %x %x\n", val, r, g, b);
-
-	return  (b << 16) + (g << 8) + r;
-    }
-
-    void draw() {
-	float  spec_index_per_px, intensity_per_dB;
-	int    px_per_sec;
-	int    index, dy, dy_blocks, bytes_in_row_of_blocks, b;
-	int    px, py, intensity;
-	unsigned *last_row, *pdest, *psrc;
-
-	/* detect resizing of window */
-
-	if ((h() != prev_h) || (w() != prev_w)) {
-	    delete pixel_buf;
-	    new_pixel_buf(w(), h());
-	}
-
-	Fl_Box::draw();
-
-	// determine dy, the height of one "block"
-
-	px_per_sec = (float)h()/WATERFALL_SECS_Y;
-	dy = DT*px_per_sec;
-
-	// number of dy high blocks in spectrogram
-
-	dy_blocks = h()/dy;
-
-	// shift previous bit map
-					       
-	bytes_in_row_of_blocks = dy*w()*sizeof(unsigned);
-
-	for(b=0; b<dy_blocks-1; b++) {
-	    pdest = pixel_buf + b*w()*dy;
-	    psrc  = pixel_buf + (b+1)*w()*dy;
-	    memcpy(pdest, psrc, bytes_in_row_of_blocks);
-	}
-
-	// create a new row of blocks at bottom
-
-	spec_index_per_px = (float)FDMDV_NSPEC/(float)w();
-	intensity_per_dB = (float)256/(MAX_DB - MIN_DB);
-	last_row = pixel_buf + dy*(dy_blocks - 1)*w();
-
-	for(px=0; px<w(); px++) {
-	    index = px*spec_index_per_px;
-	    intensity = intensity_per_dB * (av_mag[index] - MIN_DB);
-	    if (intensity > 255) intensity = 255;
-	    if (intensity < 0) intensity = 0;
-
-	    if (greyscale) {
-		for(py=0; py<dy; py++)
-		    last_row[px+py*w()] = intensity<<8;
-	    }
-	    else {
-		for(py=0; py<dy; py++)
-		    last_row[px+py*w()] = heatmap_lut[intensity];
-	    }
-	}
-
-	// update bit map
-
-	fl_draw_image((uchar*)pixel_buf, x(), y(), w(), h(), 4, 0);
-
-    }
-
-public:
-
-    Waterfall(int x, int y, int w, int h): Fl_Box(x, y, w, h, "Waterfall")
-    {
-	int   i;
-
-	for(i=0; i<255; i++) {
-	    heatmap_lut[i] = heatmap((float)i, 0.0, 255.0);
-	}
-	greyscale = 0;
-
-	align(FL_ALIGN_TOP);
-	labelsize(10);
-	new_pixel_buf(w,h);
-    };
-
-    ~Waterfall() {
-	delete pixel_buf;
-    }
-};
-
-
-class Scatter: public Fl_Box {
-protected:
-    COMP mem[SCATTER_MEM];
-    COMP new_samples[FDMDV_NSYM];
-    int  prev_w, prev_h, prev_x, prev_y;
-
-    void draw() {
-	float x_scale;
-	float y_scale;
-	int   i, j, x1, y1;
-
-	Fl_Box::draw();
-
-	/* detect resizing of window */
-
-	if ((h() != prev_h) || (w() != prev_w) || (x() != prev_x) || (y() != prev_y)) {
-	    fl_color(FL_BLACK);
-	    fl_rectf(x(),y(),w(),h());
-	    prev_h = h(); prev_w = w(); prev_x = x(); prev_y = y();
-	}
-
-	fl_push_clip(x(),y(),w(),h());
-
-	x_scale = w()/SCATTER_X_MAX;
-	y_scale = h()/SCATTER_Y_MAX;
-
-	// erase last samples
-
-	fl_color(FL_BLACK);
-	for(i=0; i<FDMDV_NSYM; i++) {
-	    x1 = x_scale * mem[i].real + x() + w()/2;
-	    y1 = y_scale * mem[i].imag + y() + h()/2;
-	    fl_point(x1, y1);
-	    mem[i] = mem[i+FDMDV_NSYM];
-	}
-
-	// shift memory
-
-	for(i=FDMDV_NSYM; i<SCATTER_MEM-FDMDV_NSYM; i++) {
-	    mem[i] = mem[i+FDMDV_NSYM];
-	}
-
-	// draw new samples
-
-	fl_color(FL_GREEN);
-	for(i=SCATTER_MEM-FDMDV_NSYM, j=0; i<SCATTER_MEM; i++,j++) {
-	    x1 = x_scale * new_samples[j].real + x() + w()/2;
-	    y1 = y_scale * new_samples[j].imag + y() + h()/2;
-	    fl_point(x1, y1);
-	    mem[i] = new_samples[j];
-	}
-	fl_pop_clip();
-    }
-
-public:
-    Scatter(int x, int y, int w, int h): Fl_Box(x, y, w, h, "Scatter")
-    {
-	int i;
-
-	align(FL_ALIGN_TOP);
-	labelsize(10);
-
-	for(i=0; i<SCATTER_MEM; i++) {
-	    mem[i].real = 0.0;
-	    mem[i].imag = 0.0;
-	}
-
-	prev_w = 0; prev_h = 0; prev_x = 0; prev_y = 0;
-    };
-
-    void add_new_samples(COMP samples[]) {
-	int i;
-
-	for(i=0; i<FDMDV_NSYM; i++)
-	    new_samples[i] = samples[i];
-    }
-
-};
-
-
-// general purpose way of plotting scalar values that are 
-// updated once per frame
-
-class Scalar: public Fl_Box {
-protected:
-    int    x_max, y_max;
-    float *mem;              /* array of x_max samples */
-    float  new_sample;
-    int    index, step;
-    int    prev_w, prev_h, prev_x, prev_y;
-
-    int clip(int y1) {
-	if (y1 > (h()/2 - 10))
-	    y1 = h()/2 - 10;       
-	if (y1 < -(h()/2 - 10))
-	    y1 = -(h()/2 - 10);       
-	return y1;
-    }
-
-    void draw() {
-	float x_scale;
-	float y_scale;
-	int   i, x1, y1, x2, y2;
-	char  label[100];
-
-	Fl_Box::draw();
-
-	/* detect resizing of window */
-
-	if ((h() != prev_h) || (w() != prev_w) || (x() != prev_x) || (y() != prev_y)) {
-	    fl_color(FL_BLACK);
-	    fl_rectf(x(),y(),w(),h());
-	    prev_h = h(); prev_w = w(); prev_x = x(); prev_y = y();
-	}
-
-	fl_push_clip(x(),y(),w(),h());
-
-	x_scale = (float)w()/x_max;
-	y_scale = (float)h()/(2.0*y_max);
-
-	// erase last sample
-
-	fl_color(FL_BLACK);
-	x1 = x_scale * index + x();
-	y1 = y_scale * mem[index];
-	y1 = clip(y1);
-	y1 = y() + h()/2 - y1;
-	fl_point(x1, y1);
-
-	// draw new sample
-
-	fl_color(FL_GREEN);
-	x1 = x_scale * index + x();
-	y1 = y_scale * new_sample;
-	y1 = clip(y1);
-	y1 = y() + h()/2 - y1;
-	fl_point(x1, y1);
-	mem[index] = new_sample;
-
-	index++;
-	if (index >=  x_max)
-	    index = 0;
-
-	// y axis graticule
-
-	step = 10;
-
-	while ((2.0*y_max/step) > 10)
-	    step *= 2.0;
-	while ((2.0*y_max/step) < 4)
-	    step /= 2.0;
-
-	fl_color(FL_DARK_GREEN);
-	fl_line_style(FL_DOT);
-	for(i=-y_max; i<y_max; i+=step) {
-	    x1 = x();
-	    y1 = y() + h()/2 - i*y_scale;
-	    x2 = x() + w();
-	    y2 = y1;
-	    fl_line(x1,y1,x2,y2);   
-	}
-
-	// y axis graticule labels
-
-	fl_color(FL_GREEN);
-	fl_line_style(FL_SOLID);
-	for(i=-y_max; i<y_max; i+=step) {
-	    x1 = x();
-	    y1 = y() + h()/2 - i*y_scale;
-	    sprintf(label, "%d", i);
-	    fl_draw(label, x1, y1);
-	}
-	fl_pop_clip();
-    }
-
-public:
-    Scalar(int x, int y, int w, int h, int x_max_, int y_max_, const char name[]): Fl_Box(x, y, w, h, name)
-    {
-	int i;
-
-	align(FL_ALIGN_TOP);
-	labelsize(10);
-
-	x_max = x_max_; y_max = y_max_;
-	mem = new float[x_max];
-	for(i=0; i<x_max; i++) {
-	    mem[i] = 0.0;
-	}
-
-	prev_w = 0; prev_h = 0; prev_x = 0; prev_y = 0;
-	index = 0;
-    };
-
-    ~Scalar() {
-	delete mem;
-    }
-
-    void add_new_sample(float sample) {
-	new_sample = sample;
-    }
-
-};
-
-
-// update average of each spectrum point
-    
-void new_data(float mag_dB[]) {
-    int i;
-
-    for(i=0; i<FDMDV_NSPEC; i++)
-	av_mag[i] = (1.0 - BETA)*av_mag[i] + BETA*mag_dB[i];
-}
-
-
-/*------------------------------------------------------------------*\
-
-  FUNCTION: per_frame_rx_processing()
-  AUTHOR..: David Rowe
-  DATE....: July 2012
-  
-  Called every rx frame to take a buffer of input modem samples and
-  convert them to a buffer of output speech samples.
-
-  The sample source could be a sound card or file.  The sample source
-  supplies a fixed number of samples with each call.  However
-  fdmdv_demod requires a variable number of samples for each call.
-  This function will buffer as appropriate and call fdmdv_demod with
-  the correct number of samples.
-
-  The processing sequence is:
-
-  collect demod input samples from sound card 1 A/D
-  while we have enough samples:
-    demod samples into bits
-    decode bits into speech samples
-    output a buffer of speech samples to sound card 2 D/A
-
-  Note that sound card 1 and sound card 2 will have slightly different
-  sample rates, as their sample clocks are not syncronised.  We
-  effectively lock the system to the demod A/D (sound card 1) sample
-  rate. This ensures the demod gets a continuous sequence of samples,
-  maintaining sync. Sample underflow or overflow will instead occur on
-  the sound card 2 D/A.  This is acceptable as a buffer of lost or
-  extra speech samples is unlikely to be noticed.
-
-  The situation is actually a little more complex than that.  Through
-  the demod timing estimation the buffers supplied to sound card D/A 2
-  are effectively clocked at the remote modulator sound card D/A clock
-  rate.  We slip/gain buffers supplied to sound card 2 to compensate.
-
-  The current demod handles varying clock rates by having a variable
-  number of input samples, e.g. 120 160 (nominal) or 200.  However the
-  A/D always delivers a fixed number of samples.
-
-  So we currently need some logic between the A/D and the demod:
-    + A/D delivers fixed number of samples
-    + demod processes a variable number of samples
-    + this means we run demod 0,1 or 2 times, depending 
-      on number of buffered A/D samples
-    + demod always outputs 1 frame of bits
-    + so run demod and speech decoder 0, 1 or 2 times
-  
-  The ouput of the demod is codec voice data so it's OK if we miss or
-  repeat a frame every now and again.
-
-  TODOs:
-
-    + this might work with arbitrary input and output buffer lengths,
-    0,1, or 2 only apply if we are inputting the nominal number of
-    samples on every call.
-
-    + so the I/O buffer sizes might not matter, as long as they of
-    reasonable size (say twice the nominal size).
-
-\*------------------------------------------------------------------*/
-
-void per_frame_rx_processing(short  output_buf[], /* output buf of decoded speech samples          */
-			     int   *n_output_buf, /* how many samples currently in output_buf[]    */
-                             int    codec_bits[], /* current frame of bits for decoder             */
-			     short  input_buf[],  /* input buf of modem samples input to demod     */ 
-			     int   *n_input_buf,  /* how many samples currently in input_buf[]     */
-			     int   *nin,          /* amount of samples demod needs for next call   */
-			     int   *state,        /* used to collect codec_bits[] halves           */
-			     struct CODEC2 *c2    /* Codec 2 states                                */
-			     )
-{
-    struct FDMDV_STATS stats;
-    int    sync_bit;
-    float  rx_fdm[FDMDV_MAX_SAMPLES_PER_FRAME];
-    int    rx_bits[FDMDV_BITS_PER_FRAME];
-    uchar  packed_bits[BYTES_PER_CODEC_FRAME];
-    float  rx_spec[FDMDV_NSPEC];
-    int    i, nin_prev, bit, byte;
-    int    next_state;
-
-    assert(*n_input_buf <= (2*FDMDV_NOM_SAMPLES_PER_FRAME));    
-   
-    /*
-      This while loop will run the demod 0, 1 (nominal) or 2 times:
-
-      0: when tx sample clock runs faster than rx, occasionally we
-         will run out of samples
-
-      1: normal, run decoder once, every 2nd frame output a frame of
-         speech samples to D/A
-
-      2: when tx sample clock runs slower than rx, occasionally we will
-         have enough samples to run demod twice.
-
-      With a +/- 10 Hz sample clock difference at FS=8000Hz (+/- 1250
-      ppm), case 0 or 1 occured about once every 30 seconds.  This is
-      no problem for the decoded audio.
-    */
-
-    while(*n_input_buf >= *nin) {
-
-	// demod per frame processing
-
-	for(i=0; i<*nin; i++)
-	    rx_fdm[i] = (float)input_buf[i]/FDMDV_SCALE;
-	nin_prev = *nin;
-	fdmdv_demod(fdmdv, rx_bits, &sync_bit, rx_fdm, nin);
-	*n_input_buf -= nin_prev;
-	assert(*n_input_buf >= 0);
-
-	// shift input buffer
-
-	for(i=0; i<*n_input_buf; i++)
-	    input_buf[i] = input_buf[i+nin_prev];
-
-	// compute rx spectrum & get demod stats, and update GUI plot data
-
-	fdmdv_get_rx_spectrum(fdmdv, rx_spec, rx_fdm, nin_prev);
-	fdmdv_get_demod_stats(fdmdv, &stats);
-	new_data(rx_spec);
-	aScatter->add_new_samples(stats.rx_symbols);
-	aTimingEst->add_new_sample(stats.rx_timing);
-	aFreqEst->add_new_sample(stats.foff);
-	aSNR->add_new_sample(stats.snr_est);
-
-	/* 
-	   State machine to:
-
-	   + Mute decoded audio when out of sync.  The demod is synced
-	     when we are using the fine freq estimate and SNR is above
-	     a thresh.
-
-	   + Decode codec bits only if we have a 0,1 sync bit
-	     sequence.  Collects two frames of demod bits to decode
-	     one frame of codec bits.
-	*/
-
-	next_state = *state;
-	switch (*state) {
-	case 0:
-	    /* mute output audio when out of sync */
-
-	    if (*n_output_buf < 2*codec2_samples_per_frame(c2) - N8) {
-		for(i=0; i<N8; i++)
-		    output_buf[*n_output_buf + i] = 0;
-		*n_output_buf += N8;
-	    }
-	    assert(*n_output_buf <= (2*codec2_samples_per_frame(c2)));  
-
-	    if ((stats.fest_coarse_fine == 1) && (stats.snr_est > 3.0))
-		next_state = 1;
-
-	    break;
-	case 1:
-	    if (sync_bit == 0) {
-		next_state = 2;
-
-		/* first half of frame of codec bits */
-
-		memcpy(codec_bits, rx_bits, FDMDV_BITS_PER_FRAME*sizeof(int));
-	    }
-	    else
-		next_state = 1;
-	    
-	    if (stats.fest_coarse_fine == 0)
-		next_state = 0;
-
-	    break;
-	case 2:
-	    next_state = 1;
-
-	    if (stats.fest_coarse_fine == 0)
-		next_state = 0;
-
-	    if (sync_bit == 1) {
-		/* second half of frame of codec bits */
-
-		memcpy(&codec_bits[FDMDV_BITS_PER_FRAME], rx_bits, FDMDV_BITS_PER_FRAME*sizeof(int));
-
-		/* pack bits, MSB received first  */
-
-		bit = 7; byte = 0;
-		memset(packed_bits, 0, BYTES_PER_CODEC_FRAME);
-		for(i=0; i<BITS_PER_CODEC_FRAME; i++) {
-		    packed_bits[byte] |= (codec_bits[i] << bit);
-		    bit--;
-		    if (bit < 0) {
-			bit = 7;
-			byte++;
-		    }
-		}
-		assert(byte == BYTES_PER_CODEC_FRAME);
-
-		/* add decoded speech to end of output buffer */
-
-		if (*n_output_buf <= codec2_samples_per_frame(c2)) {
-		    codec2_decode(c2, &output_buf[*n_output_buf], packed_bits);
-		    *n_output_buf += codec2_samples_per_frame(c2);
-		}
-		assert(*n_output_buf <= (2*codec2_samples_per_frame(c2)));  
-		
-	    }
-	    break;
-	}	
-	*state = next_state;
-    }
-}
-
-
-/* 
-   Redraw windows every DT seconds.
-*/
-
-void update_gui(int nin, float *Ts) {
-
-    *Ts += (float)nin/FS;
-	
-    *Ts += (float)nin/FS;
-    if (*Ts >= DT) {
-	*Ts -= DT;
-	if (!zoomSpectrumWindow->shown() && !zoomWaterfallWindow->shown()) {
-	    aSpectrum->redraw();
-	    aWaterfall->redraw();
-	    aScatter->redraw();
-	    aTimingEst->redraw();
-	    aFreqEst->redraw();
-	    aSNR->redraw();
-	}
-	if (zoomSpectrumWindow->shown())		
-	    aZoomedSpectrum->redraw();		
-	if (zoomWaterfallWindow->shown())		
-	    aZoomedWaterfall->redraw();		
-    }
-}
-
-
-/*
-  idle() is the FLTK function that gets continusouly called when FLTK
-  is not doing GUI work.  We use this function for providing file
-  input to update the GUI when simulating real time operation.
-*/
-
-void idle(void*) {
-    int ret, i;
-
-    if (fin_name != NULL) {
-	ret = fread(&input_buf[n_input_buf], 
-		    sizeof(short), 
-		    FDMDV_NOM_SAMPLES_PER_FRAME, 
-		    fin);
-	n_input_buf += FDMDV_NOM_SAMPLES_PER_FRAME;             
-
-	per_frame_rx_processing(output_buf, &n_output_buf,
-				codec_bits,
-				input_buf, &n_input_buf, 
-				&nin, &state, codec2);
-
-	if (fout_name != NULL) {
-	    if (n_output_buf >= N8) {
-		ret = fwrite(output_buf, sizeof(short), N8, fout);
-		n_output_buf -= N8;
-		assert(n_output_buf >= 0);
-		
-		/* shift speech sample output buffer */
-
-		for(i=0; i<n_output_buf; i++)
-		    output_buf[i] = output_buf[i+N8];
-	    }
-	}
-    }
-
-    update_gui(nin, &Ts);
-
-    // simulate time delay from real world A/D input, and pause betwen
-    // screen updates
-
-    usleep(20000);
-}
-
-
-/* 
-   This routine will be called by the PortAudio engine when audio is
-   available.
-*/
-
-static int callback( const void *inputBuffer, void *outputBuffer,
-		     unsigned long framesPerBuffer,
-		     const PaStreamCallbackTimeInfo* timeInfo,
-		     PaStreamCallbackFlags statusFlags,
-		     void *userData )
-{
-    paCallBackData *cbData = (paCallBackData*)userData;
-    uint        i;
-    short      *rptr = (short*)inputBuffer;
-    short      *wptr = (short*)outputBuffer;
-    float      *in8k = cbData->in8k;
-    float      *in48k = cbData->in48k;
-    float       out8k[N8];
-    float       out48k[N48];
-    short       out48k_short[N48];
-
-    (void) timeInfo;
-    (void) statusFlags;
-
-    assert(inputBuffer != NULL);
-
-    /* Convert input model samples from 48 to 8 kHz ------------ */
-
-    /* just use left channel */
-
-    for(i=0; i<framesPerBuffer; i++,rptr+=2)
-	in48k[i+FDMDV_OS_TAPS] = *rptr; 
-
-    /* downsample and update filter memory */
-
-    fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	in48k[i] = in48k[i+framesPerBuffer];
-
-    /* run demod, decoder and update GUI info ------------------ */
-
-    for(i=0; i<N8; i++)
-	input_buf[n_input_buf+i] = (short)out8k[i];
-    n_input_buf += FDMDV_NOM_SAMPLES_PER_FRAME;             
-
-    per_frame_rx_processing(output_buf, &n_output_buf,
-			    codec_bits,
-			    input_buf, &n_input_buf, 
-			    &nin, &state, codec2);
-
-    /* if demod out of sync copy input audio from A/D to aid in tuning */
-
-    if (n_output_buf >= N8) {
-       if (state == 0) {
-	   for(i=0; i<N8; i++)
-	       in8k[MEM8+i] = out8k[i];       /* A/D signal */
-       }
-       else {
-	   for(i=0; i<N8; i++)
-	       in8k[MEM8+i] = output_buf[i];  /* decoded spech */
-       }
-       n_output_buf -= N8;
-    }
-    assert(n_output_buf >= 0);
-
-    /* shift speech samples in output buffer */
-
-    for(i=0; i<(uint)n_output_buf; i++)
-	output_buf[i] = output_buf[i+N8];
-
-    /* Convert output speech to 48 kHz sample rate ------------- */
-
-    /* upsample and update filter memory */
-
-    fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
-    for(i=0; i<MEM8; i++)
-	in8k[i] = in8k[i+N8];
-
-    assert(outputBuffer != NULL);
-
-    /* write signal to both channels */
-
-    for(i=0; i<N48; i++)
-	out48k_short[i] = (short)out48k[i];
-    for(i=0; i<framesPerBuffer; i++,wptr+=2) {
-	wptr[0] = out48k_short[i]; 
-	wptr[1] = out48k_short[i]; 
-    }
-
-    return paContinue;
-}
-
-
-int arg_callback(int argc, char **argv, int &i) {
-    if (argv[i][1] == 'i') {
-	if ((i+1) >= argc) 
-	    return 0;
-	fin_name = argv[i+1];
-	i += 2;
-	return 2;
-    }
-    if (argv[i][1] == 'o') {
-	if ((i+1) >= argc) 
-	    return 0;
-	fout_name = argv[i+1];
-	i += 2;
-	return 2;
-    }
-    if (argv[i][1] == 's') {
-	if ((i+1) >= argc) 
-	    return 0;
-	sound_dev_name = argv[i+1];
-	i += 2;
-	return 2;
-    }
-    return 0;
-}
-
-/*------------------------------------------------------------*\
-
-                                 MAIN
-
-\*------------------------------------------------------------*/
-
-int main(int argc, char **argv) {
-    int                ret;
-    int                i;
-    PaStreamParameters inputParameters, outputParameters;
-    paCallBackData     cbData;
-
-    i = 1;
-    Fl::args(argc,argv,i,arg_callback);
-
-    if (argc == 1) {
-	printf("usage: %s [-i inputFdmdvRawFile] [-o outputRawSoundFile] [-s inputSoundDevice]\n", argv[0]);
-	exit(0);
-    }
-
-    if (fin_name != NULL) {
-	fin = fopen(fin_name,"rb");
-	if (fin == NULL) {
-	    fprintf(stderr, "Error opening input fdmdv raw file %s\n", fin_name);
-	    exit(1);
-	}
-    }
-    
-    if (fout_name != NULL) {
-	fout = fopen(fout_name,"wb");
-	if (fout == NULL) {
-	    fprintf(stderr, "Error opening output speech raw file %s\n", fout_name);
-	    exit(1);
-	}
-    }
-    
-    for(i=0; i<FDMDV_NSPEC; i++)
-	av_mag[i] = -40.0;
-
-    fdmdv = fdmdv_create();
-    codec2 = codec2_create(CODEC2_MODE_1400);
-    output_buf = (short*)malloc(2*sizeof(short)*codec2_samples_per_frame(codec2));
-
-    /*------------------------------------------------------------------------*\
-
-                           Init Sound Card I/O
-
-    \*------------------------------------------------------------------------*/
-
-    if (sound_dev_name != NULL) {
-	for(i=0; i<MEM8; i++)
-	    cbData.in8k[i] = 0.0;
-	for(i=0; i<FDMDV_OS_TAPS; i++)
-	    cbData.in48k[i] = 0.0;
-
-	err = Pa_Initialize();
-	if( err != paNoError ) goto pa_error;
-	inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
-	if (inputParameters.device == paNoDevice) {
-	    fprintf(stderr,"Error: No default input device.\n");
-	    goto pa_error;
-	}
-	inputParameters.channelCount =  NUM_CHANNELS;        /* stereo input */
-	inputParameters.sampleFormat = paInt16;
-	inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
-	inputParameters.hostApiSpecificStreamInfo = NULL;
-
-	outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
-	if (outputParameters.device == paNoDevice) {
-	    fprintf(stderr,"Error: No default output device.\n");
-	    goto pa_error;
-	}
-	outputParameters.channelCount = NUM_CHANNELS;         /* stereo output */
-	outputParameters.sampleFormat = paInt16;
-	outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-	outputParameters.hostApiSpecificStreamInfo = NULL;
-
-	err = Pa_OpenStream(
-			    &stream,
-			    &inputParameters,
-			    &outputParameters,                  
-			    SAMPLE_RATE,
-			    N48,
-			    paClipOff,      
-			    callback,
-			    &cbData);
-
-	if( err != paNoError ) goto pa_error;
-    }
-
-    /*------------------------------------------------------------------------*\
-
-                                 Init GUI
-
-    \*------------------------------------------------------------------------*/
-
-    // recommended to prevent dithering and stopped display being
-    // covered by black flickering squares
-
-    Fl::visual(FL_RGB);
-
-    // set up main window
-
-    window = new Fl_Window(W, SP+H2+SP+SP+H2+SP, "fl_fmdv");
-    //window->size_range(100, 100);
-    //window->resizable();
-    aSpectrum = new Spectrum(SP, SP, W3-2*SP, H2);
-    aWaterfall = new Waterfall(SP, SP+H2+SP+SP, W3-2*SP, H2);
-    aScatter = new Scatter(W3+SP, SP, W3-2*SP, H2);
-    aTimingEst = new Scalar(W3+SP, SP+H2+SP+SP, W3-2*SP, H2, 100, 80, "Timing Est");
-    aFreqEst = new Scalar(2*W3+SP, SP, W3-2*SP, H2, 100, 100, "Frequency Est");
-    aSNR = new Scalar(2*W3+SP, SP+H2+SP+SP, W3-2*SP, H2, 100, 20, "SNR");
-
-    Fl::add_idle(idle);
-
-    window->end();
-
-    // set up zoomed spectrum window
-
-    zoomSpectrumWindow = new Fl_Window(W, H, "Spectrum");
-    aZoomedSpectrum = new Spectrum(SP, SP, W-2*SP, H-2*SP);
-    zoomSpectrumWindow->end();
-
-    // set up zoomed waterfall window
-
-    zoomWaterfallWindow = new Fl_Window(W, H, "Waterfall");
-    aZoomedWaterfall = new Waterfall(SP, SP, W-2*SP, H-2*SP);
-    zoomWaterfallWindow->end();
-
-    if (sound_dev_name != NULL) {
-	err = Pa_StartStream( stream );
-	if( err != paNoError ) goto pa_error;
-    }
-
-    // show the main window and start running
-
-    window->show(argc, argv);
-    Fl::run();
-
-    if (sound_dev_name != NULL) {
-	err = Pa_StopStream( stream );
-	if( err != paNoError ) goto pa_error;
-	Pa_CloseStream( stream );
-	Pa_Terminate();
-    }
-
-    fdmdv_destroy(fdmdv);
-    codec2_destroy(codec2);
-    free(output_buf);
-
-    if (fin_name != NULL)
-	fclose(fin);
-    if (fout_name != NULL)
-	fclose(fout);
-
-    return ret;
-
-    // Portaudio error handling
-
-pa_error:
-    if( stream ) {
-       Pa_AbortStream( stream );
-       Pa_CloseStream( stream );
-    }
-    Pa_Terminate();
-    fprintf( stderr, "An error occured while using the portaudio stream\n" );
-    fprintf( stderr, "Error number: %d\n", err );
-    fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-    return -1;
-}
diff --git a/codec2-dev/portaudio/Makefile b/codec2-dev/portaudio/Makefile
deleted file mode 100644
index 39255b6c..00000000
--- a/codec2-dev/portaudio/Makefile
+++ /dev/null
@@ -1,23 +0,0 @@
-# Makefile for Portaudio test programs
-# Requires Portaudio V19
-
-CFLAGS =  -g -Wall -I../src
-LIBS = -lm -lportaudio -pthread
-SRC  = ../src/fdmdv.c ../src/kiss_fft.c ../src/fifo.c
-
-all: pa_rec pa_play pa_recplay pa_impresp
-
-pa_rec: Makefile pa_rec.c $(SRC)
-	gcc $(CFLAGS) pa_rec.c $(SRC) -o pa_rec $(LIBS)
-
-pa_play: Makefile pa_play.c $(SRC)
-	gcc $(CFLAGS) pa_play.c $(SRC) -o pa_play $(LIBS)
-
-pa_recplay: Makefile pa_recplay.c $(SRC)
-	gcc $(CFLAGS) pa_recplay.c $(SRC) -o pa_recplay $(LIBS)
-
-pa_impresp: Makefile pa_impresp.c $(SRC)
-	gcc $(CFLAGS) pa_impresp.c $(SRC) -o pa_impresp $(LIBS)
-
-clean:
-	rm -f pa_rec pa_play pa_recplay
diff --git a/codec2-dev/portaudio/pa_impresp.c b/codec2-dev/portaudio/pa_impresp.c
deleted file mode 100644
index 7317e07e..00000000
--- a/codec2-dev/portaudio/pa_impresp.c
+++ /dev/null
@@ -1,254 +0,0 @@
-/* 
-   pa_impresp.c
-   David Rowe
-   August 29 2012
-
-   Measures the impulse reponse of the path between the speaker and
-   microphone.  Used to explore why Codec audio quality is
-   different through a speaker and headphones.
-
-   Modified from pa_playrec.c
-*/
-
-/*
- * $Id: paex_record.c 1752 2011-09-08 03:21:55Z philburk $
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include "portaudio.h"
-#include "fdmdv.h"
-
-#define SAMPLE_RATE  48000         /* 48 kHz sampling rate rec. as we
-				      can trust accuracy of sound
-				      card */
-#define N8           160           /* processing buffer size at 8 kHz */
-#define N48          (N8*FDMDV_OS) /* processing buffer size at 48 kHz */
-#define MEM8 (FDMDV_OS_TAPS/FDMDV_OS)
-#define NUM_CHANNELS 2             /* I think most sound cards prefer
-				      stereo, we will convert to mono
-				      as we sample */
-
-#define IMPULSE_AMP    16384       /* amplitide of impulse                  */
-#define IMPULSE_PERIOD 0.1         /* period (dly between impulses) in secs */
-
-/* state information passed to call back */
-
-typedef struct {
-    float               in48k[FDMDV_OS_TAPS + N48];
-    float               in8k[MEM8 + N8];
-    FILE               *fimp;
-    float              *impulse_buf;
-    int                 impulse_buf_length;
-    int                 impulse_sample_count;
-    int                 framesLeft;
-} paTestData;
-
-
-/* 
-   This routine will be called by the PortAudio engine when audio is
-   required.  It may be called at interrupt level on some machines so
-   don't do anything that could mess up the system like calling
-   malloc() or free().
-*/
-
-static int callback( const void *inputBuffer, void *outputBuffer,
-                           unsigned long framesPerBuffer,
-                           const PaStreamCallbackTimeInfo* timeInfo,
-                           PaStreamCallbackFlags statusFlags,
-                           void *userData )
-{
-    paTestData *data = (paTestData*)userData;
-    int         i;
-    short      *rptr = (short*)inputBuffer;
-    short      *wptr = (short*)outputBuffer;
-    float      *in8k = data->in8k;
-    float      *in48k = data->in48k;
-    float       out8k[N8];
-    float       out48k[N48];
-    short       out48k_short[N48];
-    short       out8k_short[N8];
-
-    (void) timeInfo;
-    (void) statusFlags;
-
-    assert(inputBuffer != NULL);
-
-    /* just use left channel */
-
-    for(i=0; i<framesPerBuffer; i++,rptr+=2)
-	data->in48k[i+FDMDV_OS_TAPS] = *rptr; 
-
-    /* downsample and update filter memory */
-
-    fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	in48k[i] = in48k[i+framesPerBuffer];
-
-    /* write impulse response to disk */
-    
-    for(i=0; i<N8; i++)
-	out8k_short[i] = out8k[i];
-    fwrite(out8k_short, sizeof(short), N8, data->fimp);
-
-    /* play side, read from impulse buffer */
-
-    for(i=0; i<N8; i++) {
-	in8k[MEM8+i] = data->impulse_buf[data->impulse_sample_count];
-	data->impulse_sample_count++;
-	if (data->impulse_sample_count == data->impulse_buf_length)
-	    data->impulse_sample_count = 0;
-    }
-
-    /* upsample and update filter memory */
-
-    fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
-    for(i=0; i<MEM8; i++)
-	in8k[i] = in8k[i+N8];
-
-    assert(outputBuffer != NULL);
-
-    /* write signal to both channels */
-
-    for(i=0; i<N48; i++)
-	out48k_short[i] = (short)out48k[i];
-    for(i=0; i<framesPerBuffer; i++,wptr+=2) {
-	wptr[0] = out48k_short[i]; 
-	wptr[1] = out48k_short[i]; 
-    }
-
-    data->framesLeft -= framesPerBuffer;
-    if (data->framesLeft > 0)
-	return paContinue;
-    else
-	return paComplete;
-}
-
-int main(int argc, char *argv[])
-{
-    PaStreamParameters  inputParameters, outputParameters;
-    PaStream*           stream;
-    PaError             err = paNoError;
-    paTestData          data;
-    int                 i, numSecs;
-
-    if (argc != 3) {
-	printf("usage: %s impulseRawFile time(s)\n", argv[0]);
-	exit(0);
-    }
-
-    data.fimp = fopen(argv[1], "wb");
-    if (data.fimp == NULL) {
-	printf("Error opening impulse output file %s\n", argv[1]);
-	exit(1);
-    }
-
-    numSecs = atoi(argv[2]);
-    data.framesLeft = numSecs * SAMPLE_RATE;
-
-    /* init filter states */
-
-    for(i=0; i<MEM8; i++)
-	data.in8k[i] = 0.0;
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	data.in48k[i] = 0.0;
-
-    /* init imupulse */
-
-    data.impulse_buf_length = IMPULSE_PERIOD*(SAMPLE_RATE/FDMDV_OS);
-    printf("%d\n",data.impulse_buf_length);
-    data.impulse_buf = (float*)malloc(data.impulse_buf_length*sizeof(float));
-    assert(data.impulse_buf != NULL);
-    data.impulse_buf[0] = IMPULSE_AMP;
-    for(i=1; i<data.impulse_buf_length; i++)
-	data.impulse_buf[i] = 0;
-    data.impulse_sample_count = 0;
-
-    err = Pa_Initialize();
-    if( err != paNoError ) goto done;
-
-    inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
-    if (inputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default input device.\n");
-        goto done;
-    }
-    inputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
-    inputParameters.sampleFormat = paInt16;
-    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
-    inputParameters.hostApiSpecificStreamInfo = NULL;
-
-    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
-    if (outputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default output device.\n");
-        goto done;
-    }
-    outputParameters.channelCount = NUM_CHANNELS;         /* stereo output */
-    outputParameters.sampleFormat = paInt16;
-    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-    outputParameters.hostApiSpecificStreamInfo = NULL;
-
-    /* Play some audio --------------------------------------------- */
-
-    err = Pa_OpenStream(
-              &stream,
-	      &inputParameters,
-              &outputParameters,
-              SAMPLE_RATE,
-              N48,
-              paClipOff,      
-              callback,
-              &data );
-    if( err != paNoError ) goto done;
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto done;
-
-    while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
-    {
-        Pa_Sleep(100);
-    }
-    if( err < 0 ) goto done;
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto done;
-
-
-done:
-    Pa_Terminate();
-    if( err != paNoError )
-    {
-        fprintf( stderr, "An error occured while using the portaudio stream\n" );
-        fprintf( stderr, "Error number: %d\n", err );
-        fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-        err = 1;          /* Always return 0 or 1, but no other return codes. */
-    }
-
-    fclose(data.fimp);
-
-    return err;
-}
-
diff --git a/codec2-dev/portaudio/pa_play.c b/codec2-dev/portaudio/pa_play.c
deleted file mode 100644
index 00ea0868..00000000
--- a/codec2-dev/portaudio/pa_play.c
+++ /dev/null
@@ -1,199 +0,0 @@
-/* 
-   pa_play.c
-   David Rowe
-   July 8 2012
-
-   Converts samples from a 16 bit short 8000 Hz rawfile to 480000Hz
-   sample rate and plays them using the default sound device.  Used as
-   an intermediate step in Portaudio integration.
-
-   Modified from paex_record.c Portaudio example. Original author
-   author Phil Burk http://www.softsynth.com
-*/
-
-/*
- * $Id: paex_record.c 1752 2011-09-08 03:21:55Z philburk $
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include "portaudio.h"
-#include "fdmdv.h"
-
-#define SAMPLE_RATE  48000         /* 48 kHz sampling rate rec. as we
-				      can trust accuracy of sound
-				      card */
-#define N8           160           /* processing buffer size at 8 kHz */
-#define N48          (N8*FDMDV_OS) /* processing buffer size at 48 kHz */
-#define MEM8 (FDMDV_OS_TAPS/FDMDV_OS)
-#define NUM_CHANNELS 2             /* I think most sound cards prefer
-				      stereo, we will convert to mono
-				      as we sample */
-
-/* state information passed to call back */
-
-typedef struct {
-    FILE               *fin;
-    float               in8k[MEM8 + N8];
-} paTestData;
-
-
-/* 
-   This routine will be called by the PortAudio engine when audio is
-   required.  It may be called at interrupt level on some machines so
-   don't do anything that could mess up the system like calling
-   malloc() or free().
-*/
-
-static int playCallback( const void *inputBuffer, void *outputBuffer,
-                           unsigned long framesPerBuffer,
-                           const PaStreamCallbackTimeInfo* timeInfo,
-                           PaStreamCallbackFlags statusFlags,
-                           void *userData )
-{
-    paTestData *data = (paTestData*)userData;
-    FILE       *fin = data->fin;
-    int         i, nread;
-    int         finished;
-    short      *wptr = (short*)outputBuffer;
-    float      *in8k = data->in8k;
-    float       out48k[N48];
-    short       out48k_short[N48];
-    short       in8k_short[N8];
-
-    (void) outputBuffer; /* Prevent unused variable warnings. */
-    (void) timeInfo;
-    (void) statusFlags;
-    (void) userData;
-
-    /* note Portaudio docs recs. against making systems calls like
-       fwrite() in this callback but seems to work OK */
-    
-    nread = fread(in8k_short, sizeof(short), N8, fin);
-    if (nread == N8)
-	finished = paContinue;
-    else
-	finished = paComplete;
-
-    for(i=0; i<N8; i++)
-	in8k[MEM8+i] = in8k_short[i];
-
-    /* upsample and update filter memory */
-
-    fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
-    for(i=0; i<MEM8; i++)
-	in8k[i] = in8k[i+N8];
-
-    assert(outputBuffer != NULL);
-
-    /* write signal to both channels */
-
-    for(i=0; i<N48; i++)
-	out48k_short[i] = (short)out48k[i];
-    for(i=0; i<framesPerBuffer; i++,wptr+=2) {
-	wptr[0] = out48k_short[i]; 
-	wptr[1] = out48k_short[i]; 
-    }
-
-    return finished;
-}
-
-int main(int argc, char *argv[])
-{
-    PaStreamParameters  outputParameters;
-    PaStream*           stream;
-    PaError             err = paNoError;
-    paTestData          data;
-    int                 i;
-
-    if (argc != 2) {
-	printf("usage: %s rawFile\n", argv[0]);
-	exit(0);
-    }
-
-    data.fin = fopen(argv[1], "rt");
-    if (data.fin == NULL) {
-	printf("Error opening input raw file %s\n", argv[1]);
-	exit(1);
-    }
-
-    for(i=0; i<MEM8; i++)
-	data.in8k[i] = 0.0;
-
-    err = Pa_Initialize();
-    if( err != paNoError ) goto done;
-
-    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default input device */
-    if (outputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default output device.\n");
-        goto done;
-    }
-    outputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
-    outputParameters.sampleFormat = paInt16;
-    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-    outputParameters.hostApiSpecificStreamInfo = NULL;
-
-    /* Play some audio --------------------------------------------- */
-
-    err = Pa_OpenStream(
-              &stream,
-	      NULL,
-              &outputParameters,
-              SAMPLE_RATE,
-              N48,
-              paClipOff,      
-              playCallback,
-              &data );
-    if( err != paNoError ) goto done;
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto done;
-
-    while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
-    {
-        Pa_Sleep(100);
-    }
-    if( err < 0 ) goto done;
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto done;
-
-    fclose(data.fin);
-
-
-done:
-    Pa_Terminate();
-    if( err != paNoError )
-    {
-        fprintf( stderr, "An error occured while using the portaudio stream\n" );
-        fprintf( stderr, "Error number: %d\n", err );
-        fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-        err = 1;          /* Always return 0 or 1, but no other return codes. */
-    }
-    return err;
-}
-
diff --git a/codec2-dev/portaudio/pa_rec.c b/codec2-dev/portaudio/pa_rec.c
deleted file mode 100644
index 00023b02..00000000
--- a/codec2-dev/portaudio/pa_rec.c
+++ /dev/null
@@ -1,212 +0,0 @@
-/* 
-   pa_rec.c
-   David Rowe
-   July 6 2012
-
-   Records at 48000 Hz from default sound device, convertes to 8 kHz,
-   and saves to raw file.  Used to get experience with Portaudio.
-
-   Modified from paex_record.c Portaudio example. Original author
-   author Phil Burk http://www.softsynth.com
-
-   To Build:
-
-     gcc paex_rec.c -o paex_rec -lm -lrt -lportaudio -pthread
-*/
-
-/*
- * $Id: paex_record.c 1752 2011-09-08 03:21:55Z philburk $
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include "portaudio.h"
-#include "fdmdv.h"
-
-#define SAMPLE_RATE  48000         /* 48 kHz sampling rate rec. as we
-				      can trust accuracy of sound
-				      card */
-#define N8           160           /* processing buffer size at 8 kHz */
-#define N48          (N8*FDMDV_OS) /* processing buffer size at 48 kHz */
-#define NUM_CHANNELS 2             /* I think most sound cards prefer
-				      stereo, we will convert to mono
-				      as we sample */
-
-/* state information passed to call back */
-
-typedef struct {
-    FILE               *fout;
-    int                 framesLeft;
-    float               in48k[FDMDV_OS_TAPS + N48];
-} paTestData;
-
-
-/* 
-   This routine will be called by the PortAudio engine when audio is
-   available.  It may be called at interrupt level on some machines so
-   don't do anything that could mess up the system like calling
-   malloc() or free().
-*/
-
-static int recordCallback( const void *inputBuffer, void *outputBuffer,
-                           unsigned long framesPerBuffer,
-                           const PaStreamCallbackTimeInfo* timeInfo,
-                           PaStreamCallbackFlags statusFlags,
-                           void *userData )
-{
-    paTestData *data = (paTestData*)userData;
-    FILE       *fout = data->fout;
-    int         framesToCopy;
-    int         i;
-    int         finished;
-    short      *rptr = (short*)inputBuffer;
-    float       out8k[N8];
-    short       out8k_short[N8];
-
-    (void) outputBuffer; /* Prevent unused variable warnings. */
-    (void) timeInfo;
-    (void) statusFlags;
-    (void) userData;
-
-    if (data->framesLeft < framesPerBuffer) {
-        framesToCopy = data->framesLeft;
-        finished = paComplete;
-    } 
-    else {
-        framesToCopy = framesPerBuffer;
-        finished = paContinue;
-    }
-    data->framesLeft -= framesToCopy;
-
-    assert(inputBuffer != NULL);
-
-    /* just use left channel */
-
-    for(i=0; i<framesToCopy; i++,rptr+=2)
-	data->in48k[i+FDMDV_OS_TAPS] = *rptr; 
-
-    /* downsample and update filter memory */
-
-    fdmdv_48_to_8(out8k, &data->in48k[FDMDV_OS_TAPS], N8);
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	data->in48k[i] = data->in48k[i+framesToCopy];
-
-    /* save 8k to disk  */
-
-    for(i=0; i<N8; i++)
-	out8k_short[i] = (short)out8k[i];
-
-    /* note Portaudio docs recs. against making systems calls like
-       fwrite() in this callback but seems to work OK */
-    
-    fwrite(out8k_short, sizeof(short), N8, fout);
-
-    return finished;
-}
-
-int main(int argc, char *argv[])
-{
-    PaStreamParameters  inputParameters;
-    PaStream*           stream;
-    PaError             err = paNoError;
-    paTestData          data;
-    int                 i;
-    int                 numSecs;
-
-    if (argc != 3) {
-	printf("usage: %s rawFile time(s)\n", argv[0]);
-	exit(0);
-    }
-
-    data.fout = fopen(argv[1], "wt");
-    if (data.fout == NULL) {
-	printf("Error opening output raw file %s\n", argv[1]);
-	exit(1);
-    }
-
-    numSecs = atoi(argv[2]);
-    data.framesLeft = numSecs * SAMPLE_RATE;
-
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	data.in48k[i] = 0.0;
-
-    err = Pa_Initialize();
-    if( err != paNoError ) goto done;
-
-    printf( "PortAudio version number = %d\nPortAudio version text = '%s'\n",
-            Pa_GetVersion(), Pa_GetVersionText() );
-
-    inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
-    if (inputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default input device.\n");
-        goto done;
-    }
-    inputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
-    inputParameters.sampleFormat = paInt16;
-    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
-    inputParameters.hostApiSpecificStreamInfo = NULL;
-
-    /* Record some audio --------------------------------------------- */
-
-    err = Pa_OpenStream(
-              &stream,
-              &inputParameters,
-              NULL,                  /* &outputParameters, */
-              SAMPLE_RATE,
-              N48,
-              paClipOff,      /* we won't output out of range samples so don't bother clipping them */
-              recordCallback,
-              &data );
-    if( err != paNoError ) goto done;
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto done;
-
-    while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
-    {
-        Pa_Sleep(100);
-    }
-    if( err < 0 ) goto done;
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto done;
-
-    fclose(data.fout);
-
-
-done:
-    Pa_Terminate();
-    if( err != paNoError )
-    {
-        fprintf( stderr, "An error occured while using the portaudio stream\n" );
-        fprintf( stderr, "Error number: %d\n", err );
-        fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-        err = 1;          /* Always return 0 or 1, but no other return codes. */
-    }
-    return err;
-}
-
diff --git a/codec2-dev/portaudio/pa_recplay.c b/codec2-dev/portaudio/pa_recplay.c
deleted file mode 100644
index 6bfcd08b..00000000
--- a/codec2-dev/portaudio/pa_recplay.c
+++ /dev/null
@@ -1,270 +0,0 @@
-/* 
-   pa_recplay.c
-   David Rowe
-   July 8 2012
-
-   Echos audio from sound card input to sound card output. Samples at
-   48 kHz, converts to 8 kHz, converts back to 48kHz, and plays using
-   the default sound device.  Used as an intermediate step in
-   Portaudio integration.
-
-   Modified from paex_record.c Portaudio example. Original author
-   author Phil Burk http://www.softsynth.com
-*/
-
-/*
- * $Id: paex_record.c 1752 2011-09-08 03:21:55Z philburk $
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include "portaudio.h"
-#include "fdmdv.h"
-#include "fifo.h"
-
-#define SAMPLE_RATE  48000         /* 48 kHz sampling rate rec. as we
-				      can trust accuracy of sound
-				      card */
-#define N8           160           /* processing buffer size at 8 kHz */
-#define N48          (N8*FDMDV_OS) /* processing buffer size at 48 kHz */
-#define MEM8 (FDMDV_OS_TAPS/FDMDV_OS)
-#define NUM_CHANNELS 2             /* I think most sound cards prefer
-				      stereo, we will convert to mono
-				      as we sample */
-#define MAX_FPB      2048          /* maximum value of framesPerBuffer */
-
-/* state information passed to call back */
-
-typedef struct {
-    float               in48k[FDMDV_OS_TAPS + N48];
-    float               in8k[MEM8 + N8];
-    struct FIFO        *infifo;
-    struct FIFO        *outfifo;
-} paTestData;
-
-
-/* 
-   This routine will be called by the PortAudio engine when audio is
-   required.  It may be called at interrupt level on some machines so
-   don't do anything that could mess up the system like calling
-   malloc() or free().
-*/
-
-static int callback( const void *inputBuffer, void *outputBuffer,
-                           unsigned long framesPerBuffer,
-                           const PaStreamCallbackTimeInfo* timeInfo,
-                           PaStreamCallbackFlags statusFlags,
-                           void *userData )
-{
-    paTestData *data = (paTestData*)userData;
-    int         i;
-    short      *rptr = (short*)inputBuffer;
-    short      *wptr = (short*)outputBuffer;
-    float      *in8k = data->in8k;
-    float      *in48k = data->in48k;
-    float       out8k[N8];
-    float       out48k[N48];
-    short       out48k_short[N48];
-    short       in48k_short[N48];
-    short       indata[MAX_FPB];
-    short       outdata[MAX_FPB];
-
-    (void) timeInfo;
-    (void) statusFlags;
-
-    assert(inputBuffer != NULL);
-    assert(outputBuffer != NULL);
-
-    /* 
-       framesPerBuffer is portaudio-speak for number of samples we
-       actually get from the record side and need to provide to the
-       play side. On Linux (at least) it was found that
-       framesPerBuffer may not always be what we ask for in the
-       framesPerBuffer field of Pa_OpenStream.  For example a request
-       for 960 sample buffers lead to framesPerBuffer = 1024.
-
-       To perform the 48 to 8 kHz conversion we need an integer
-       multiple of FDMDV_OS samples to support the interpolation and
-       decimation.  As we can't guarantee the size of framesPerBuffer
-       we do a little FIFO buffering.
-    */
-
-    //printf("framesPerBuffer: %d N48 %d\n", framesPerBuffer, N48);
-
-    /* assemble a mono buffer (just use left channel) and write to FIFO */
-
-    assert(framesPerBuffer < MAX_FPB);
-    for(i=0; i<framesPerBuffer; i++,rptr+=2)
-	indata[i] = *rptr;
-    fifo_write(data->infifo, indata, framesPerBuffer);
-
-    /* while we have enough samples available ... */
-
-    //printf("infifo before: %d\n", fifo_n(data->infifo));
-    while (fifo_read(data->infifo, in48k_short, N48) == 0) {
-
-	/* convert to float */
-
-	for(i=0; i<N48; i++)
-	    in48k[FDMDV_OS_TAPS + i] = in48k_short[i];
-
-	/* downsample and update filter memory */
-
-	fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
-	for(i=0; i<FDMDV_OS_TAPS; i++)
-	    in48k[i] = in48k[i+N48];
-
-	/* play side, back up to 8k */
-
-	for(i=0; i<N8; i++)
-	    in8k[MEM8+i] = out8k[i];
-
-	/* upsample and update filter memory */
-
-	fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
-	for(i=0; i<MEM8; i++)
-	    in8k[i] = in8k[i+N8];
-
-	/* write signal to both channels */
-
-	for(i=0; i<N48; i++)
-	    out48k_short[i] = (short)out48k[i];
-
-	fifo_write(data->outfifo, out48k_short, N48);
-    }
-    //printf("infifo after: %d\n", fifo_n(data->infifo));
-    //printf("outfifo     : %d\n", fifo_n(data->outfifo));
-
-
-    /* OK now set up output samples */
-
-    if (fifo_read(data->outfifo, outdata, framesPerBuffer) == 0) {
-
-	/* write signal to both channels */
-
-	for(i=0; i<framesPerBuffer; i++,wptr+=2) {
-	    wptr[0] = outdata[i]; 
-	    wptr[1] = outdata[i]; 
-	}
-    }
-    else {
-	//printf("no data\n");
-	/* zero output if no data available */
-	for(i=0; i<framesPerBuffer; i++,wptr+=2) {
-	    wptr[0] = 0; 
-	    wptr[1] = 0; 
-	}
-    }
-
-    return paContinue;
-}
-
-int main(int argc, char *argv[])
-{
-    PaStreamParameters  inputParameters, outputParameters;
-    PaStream*           stream;
-    PaError             err = paNoError;
-    paTestData          data;
-    int                 i;
-
-    /* init callback data */
-
-    for(i=0; i<MEM8; i++)
-	data.in8k[i] = 0.0;
-    for(i=0; i<FDMDV_OS_TAPS; i++)
-	data.in48k[i] = 0.0;
-
-    data.infifo = fifo_create(2*N48);
-    data.outfifo = fifo_create(2*N48);
-
-    /* init port audio */
-
-    err = Pa_Initialize();
-    if( err != paNoError ) goto done;
-
-    inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
-    if (inputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default input device.\n");
-        goto done;
-    }
-    inputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
-    inputParameters.sampleFormat = paInt16;
-    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
-    inputParameters.hostApiSpecificStreamInfo = NULL;
-
-    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
-    if (outputParameters.device == paNoDevice) {
-        fprintf(stderr,"Error: No default output device.\n");
-        goto done;
-    }
-    outputParameters.channelCount = NUM_CHANNELS;         /* stereo output */
-    outputParameters.sampleFormat = paInt16;
-    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
-    outputParameters.hostApiSpecificStreamInfo = NULL;
-
-    /* Play some audio --------------------------------------------- */
-
-    err = Pa_OpenStream(
-              &stream,
-	      &inputParameters,
-              &outputParameters,
-              SAMPLE_RATE,
-              512,
-              paClipOff,      
-              callback,
-              &data );
-    if( err != paNoError ) goto done;
-
-    err = Pa_StartStream( stream );
-    if( err != paNoError ) goto done;
-
-    while( ( err = Pa_IsStreamActive( stream ) ) == 1 )
-    {
-        Pa_Sleep(100);
-    }
-    if( err < 0 ) goto done;
-
-    err = Pa_CloseStream( stream );
-    if( err != paNoError ) goto done;
-
-
-done:
-    Pa_Terminate();
-    if( err != paNoError )
-    {
-        fprintf( stderr, "An error occured while using the portaudio stream\n" );
-        fprintf( stderr, "Error number: %d\n", err );
-        fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
-        err = 1;          /* Always return 0 or 1, but no other return codes. */
-    }
-
-    fifo_destroy(data.infifo);
-    fifo_destroy(data.outfifo);
-
-    return err;
-}
-
diff --git a/codec2-dev/raw/sine1k_2Hz_spread.raw b/codec2-dev/raw/sine1k_2Hz_spread.raw
index 2ab521fc..1b8ad9ca 100644
Binary files a/codec2-dev/raw/sine1k_2Hz_spread.raw and b/codec2-dev/raw/sine1k_2Hz_spread.raw differ
diff --git a/codec2-dev/raw/sine1k_2ms_delay_2Hz_spread.raw b/codec2-dev/raw/sine1k_2ms_delay_2Hz_spread.raw
index d694d23f..4d3e09ec 100644
Binary files a/codec2-dev/raw/sine1k_2ms_delay_2Hz_spread.raw and b/codec2-dev/raw/sine1k_2ms_delay_2Hz_spread.raw differ
diff --git a/codec2-dev/voicing/hts1a.v b/codec2-dev/voicing/hts1a.v
deleted file mode 100644
index aedcc3e5..00000000
--- a/codec2-dev/voicing/hts1a.v
+++ /dev/null
@@ -1,299 +0,0 @@
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
diff --git a/codec2-dev/voicing/hts2a.v b/codec2-dev/voicing/hts2a.v
deleted file mode 100644
index fea4e54e..00000000
--- a/codec2-dev/voicing/hts2a.v
+++ /dev/null
@@ -1,300 +0,0 @@
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
diff --git a/codec2-dev/voicing/mmt1.v b/codec2-dev/voicing/mmt1.v
deleted file mode 100644
index a60d3412..00000000
--- a/codec2-dev/voicing/mmt1.v
+++ /dev/null
@@ -1,400 +0,0 @@
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1  
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
diff --git a/codec2-dev/voicing/morig.v b/codec2-dev/voicing/morig.v
deleted file mode 100644
index b89a925c..00000000
--- a/codec2-dev/voicing/morig.v
+++ /dev/null
@@ -1,200 +0,0 @@
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-1
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
-0
diff --git a/codec2-dev/win32/Makefile b/codec2-dev/win32/Makefile
deleted file mode 100644
index e13f8641..00000000
--- a/codec2-dev/win32/Makefile
+++ /dev/null
@@ -1,57 +0,0 @@
-# Win32 Makefile derived from the fine work of Peter Lawrence:
-#   http://code.google.com/p/codec2demo/
-#
-# David Rowe June 2012
-#
-# This makefile assumes the Linux version has been compiled first to generate
-# the machine-generated codebook source files.  It is designed to be cross compiled on
-# Linux.  
-# 
-
-# uncomment under Windows
-#CC_PREFIX = 
-# uncomment under Linux (to cross-compile for Windows)
-CC_PREFIX = i586-mingw32msvc-
-
-DLLNAME = codec2.dll
-LIBNAME = codec2.lib
-
-CFLAGS  = -O2 -Wall -D__CODEC2_WIN32__
-
-HDRS = ../src/codec2.h ../src/codec2_fdmdv.h
-
-OBJS = dump.o lpc.o nlp.o postfilter.o sine.o codec2.o fdmdv.o kiss_fft.o interp.o lsp.o phase.o quantise.o pack.o codebook.o codebookd.o codebookvq.o codebookjnd.o codebookjvm.o codebookdt.o codebookge.o
-
-all: $(DLLNAME) c2demo.exe c2enc.exe c2dec.exe fdmdv_get_test_bits.exe fdmdv_mod.exe fdmdv_demod.exe fdmdv_put_test_bits.exe
-
-$(DLLNAME): $(OBJS)
-	$(CC_PREFIX)gcc -shared -o $(DLLNAME) $(OBJS) -Wl,--out-implib,$(LIBNAME)
-
-%.o: ../src/%.c $(HDRS) Makefile
-	$(CC_PREFIX)gcc $(CFLAGS) -D__CODEC2_BUILDING_DLL__ -c $< -o $@
-
-c2demo.exe: ../src/c2demo.c  $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-c2enc.exe: ../src/c2enc.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-c2dec.exe: ../src/c2dec.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-fdmdv_get_test_bits.exe: ../src/fdmdv_get_test_bits.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-fdmdv_mod.exe: ../src/fdmdv_mod.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-fdmdv_demod.exe: ../src/fdmdv_demod.c ../src/octave.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< ../src/octave.c -o $@ $(LIBNAME)
-
-fdmdv_put_test_bits.exe: ../src/fdmdv_put_test_bits.c $(DLLNAME) $(LIBNAME)
-	$(CC_PREFIX)gcc $(CFLAGS) $< -o $@ $(LIBNAME)
-
-clean:
-	rm -f $(DLLNAME) $(LIBNAME)
-	rm -f *.o *.exe
-
diff --git a/codec2-dev/win32/c2dec.exe b/codec2-dev/win32/c2dec.exe
deleted file mode 100755
index 94528174..00000000
Binary files a/codec2-dev/win32/c2dec.exe and /dev/null differ
diff --git a/codec2-dev/win32/c2demo.exe b/codec2-dev/win32/c2demo.exe
deleted file mode 100755
index 8caf2cf4..00000000
Binary files a/codec2-dev/win32/c2demo.exe and /dev/null differ
diff --git a/codec2-dev/win32/c2enc.exe b/codec2-dev/win32/c2enc.exe
deleted file mode 100755
index e1e0b342..00000000
Binary files a/codec2-dev/win32/c2enc.exe and /dev/null differ
diff --git a/codec2-dev/win32/codec2.dll b/codec2-dev/win32/codec2.dll
deleted file mode 100755
index bab3d45c..00000000
Binary files a/codec2-dev/win32/codec2.dll and /dev/null differ
diff --git a/codec2-dev/win32/codec2.lib b/codec2-dev/win32/codec2.lib
deleted file mode 100644
index 639dd801..00000000
Binary files a/codec2-dev/win32/codec2.lib and /dev/null differ