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libs/libcodec2/README_fdmdv.txt

+
+README_fdmdv.txt
+David Rowe
+Created March 2012
+
+Introduction
+------------
+
+A 1400 bit/s Frequency Division Multiplexed Digital Voice (FDMDV) modem
+based on [1].  Used for digital audio over HF SSB.
+                     
+The FDMDV modem was first implemented in GNU Octave, then ported to C.
+Algorithm development is generally easier in Octave, but for real time
+work we need the C version.  Automated units tests ensure the
+operation of the Octave and C versions are identical.
+
+Quickstart
+----------
+
+$ cd codec2-dev
+$ ./configure && make
+$ cd src
+
+1. Generate some test bits and modulate them:
+
+    $ ./fdmdv_get_test_bits test.c2 1400
+    $ ./fdmdv_mod test.c2 test.raw
+    $ play -r 8000 -s -2 test.raw
+    
+2. Two seconds of test frame data modulated and sent out of sound device: 
+    
+    $ ./fdmdv_get_test_bits - 2800 | ./fdmdv_mod - - | play -t raw -r 8000 -s -2 -
+ 
+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 -
+
+    Use Octave to look at plots of 1 second (1400 bits) of modem operation:
+
+    $ cd ../octave
+    $ octave
+    octave:1> fdmdv_demod_c("../src/demod_dump.txt",1400)
+
+4. Run Octave simulation of entire modem and AWGN channel:
+
+    $ cd ../octave
+    $ octave
+    octave:1> fdmdv_ut
+
+5. NOTE: If you would like to play modem samples over the air please
+   convert the 8 kHz samples to 48 kHz.  Many PC sound cards have
+   wildly inaccurate sample clock rates when set to 8 kHz, but seem to
+   perform OK when set for 48 kHz.  If playing and recording files you
+   can use the sox utility:
+
+   $ sox -r 8000 -s -2 modem_sample_8kHz.raw -r 48000 modem_sample_48kHz.wav
+
+   For real time applications, the fdmdv.[ch] library includes functions to
+   convert between 48 and 8 kHz sample rates.
+
+References
+----------
+
+[1] http://n1su.com/fdmdv/FDMDV_Docs_Rel_1_4b.pdf
+[2] http://n1su.com/fdmdv/
+[3] http://www.rowetel.com/blog/?p=2433 "Testing a FDMDV Modem"
+[4] http://www.rowetel.com/blog/?p=2458 "FDMDV Modem Page" on David's web site
+
+C Code
+------
+
+src/fdmdv_mod.c - C version of modulator that takes a file of bits and
+                  converts it to a raw file of modulated samples.
+
+src/fdmdv_demod.c - C version of demodulator that takes a raw file of
+                    modulated samples and outputs a file of bits.
+                    Optionally dumps demod states to a text file which
+                    can be plotted using the Octave script
+                    fdmdv_demod_c.m
+
+src/fdmdv.h - Header file that exposes FDMDV C API functions.  Include
+              this file in your application program.
+
+src/fdmdv.c - C functions that implement the FDMDV modem.
+
+src/fdmdv-internal.h - internal states and constants for FDMDV modem,
+                       shouldn't be exposed to application program.
+
+
+unittest/tfdmdv.c - Used to conjunction with unittest/tfdmdv.m to
+                    automatically test C FDMDV functions against
+                    Octave versions.
+
+Octave Scripts
+--------------
+
+(Note these require some Octave packages to be installed, see
+octave/README.txt).
+
+fdmdv.m - Functions and variables that implement the Octave version of
+          the FDMDV modem.
+
+fdmdv_ut.m - Unit test for fdmdv Octave code, useful while
+             developing algorithm.  Includes tx/rx plus basic channel
+             simulation.
+
+             Typical run:
+
+               octave:6> fdmdv_ut
+               Eb/No (meas): 7.30 (8.29) dB
+               bits........: 2464
+               errors......: 20
+               BER.........: 0.0081
+               PAPR........: 13.54 dB
+               SNR.........: 4.0 dB
+
+               It also outputs lots of nice plots that show the
+	       operation of the modem.
+
+               For a 1400 bit/s DQPSK modem we expect about 1% BER for
+               Eb/No = 7.3dB, which corresponds to SNR = 4dB (3kHz
+               noise BW). The extra dB of measured power is due to the
+               DBPSK pilot. Currently the noise generation code
+               doesn't take the pilot power into account, so in this
+               example the real SNR is actually 5dB.
+             
+fdmdv_mod.m - Octave version of modulator that outputs a raw file.
+              The modulator is driven by a test frame of bits.  This
+              can then be played over a real channel or through a
+              channel simulator like PathSim.  The sample rate can be
+              changed using "sox" to simulate differences in tx/rx
+              sample clocks.
+
+	      To generate 10 seconds of modulated signal:
+                
+                octave:8> fdmdv_mod("test.raw",1400*10);
+
+fdmdv_demod.m - Demodulator program that takes a raw file as input,
+                and works out the bit error rate using the known test
+                frame.  Can be used to test the demod performs with
+                off-air signals, or signals that have been passed
+                through a channel simulator.
+
+		To demodulate 2 seconds of the test.raw file generated
+		above:
+
+                octave:9> fdmdv_demod("test.raw",1400*2);
+                2464 bits  0 errors  BER: 0.0000
+
+                It also produces several plots showing the internal
+		states of the demod.  Useful for debugging and
+		observing what happens with various channels.
+
+fdmdv_demod_c.m - Takes an output text file from the C demod
+                  fdmdv_demod.c and produces plots and measures BER.
+                  Useful for evaluating fdmdv_demod.c performance.
+                  The plots produced are identical to the Octave
+                  version fdmdv_demod.m, allowing direct comparison of
+                  the C and Octave versions.
+
+tfdmdv.m - Automatic tests that compare the Octave and C versions of
+           the FDMDV modem functions.  First run unittest/tfdmdv, this
+           will generate a text file with test vectors from the C
+           version.  Then run the Octave script tfdmdv and it will
+           generate Octave versions of the test vectors and compare
+           each vector with the C equivalent.  Its plots the vectors
+           and and errors (green).  Its also produces an automatic
+           check list based on test results.  If the Octave or C modem
+           code is changed, this script should be used to ensure the
+           C and Octave versions remain identical.
+
+Modelling sample clock errors using sox
+---------------------------------------
+
+This introduces a simulated 1000ppm error:
+
+  sox -r 8000 -s -2 mod_dqpsk.raw -s -2 mod_dqpsk_8008hz.raw rate -h 8008
+
+TODO
+----
+
+[ ] implement ppm measurements in fdmdv_get_demod_stats()
+[ ] try interfering sine wave
+    + maybe swept
+    + does modem fall over?
+[ ] try non-flat channel, e.g. 3dB difference between hi and low tones
+    + make sure all estimators keep working
+[ ] test rx level sensitivity, i.e. 0 to 20dB attenuation
+[ ] make fine freq indep of amplitude
+    + use angle rather than imag coord
+[ ] document use of fdmdv_ut and fdmdv_demod + PathSim
+[ ] more positive form of sync reqd for DV frames?
+    + like using coarse_fine==1 to decode valid DV frame bit?
+    + when should we start decoding?
+[ ] more robust track/acquite state machine?
+    + e.g. hang on thru the fades?
+[ ] PAPR idea
+    + automatically tweak phases to reduce PAPR, e.g. slow variations in freq...
+[ ] why is pilot noise_est twice as big as other carriers

libs/libcodec2/asterisk/README

+README for codec2/asterisk
+Asterisk Codec 2 support
+
+Test Configuration
+------------------
+
+Codec 2 is used to trunk calls between two Asterisk boxes:
+
+    A - SIP phone - Asterisk A - Codec2 - Asterisk B - SIP Phone - B
+
+The two SIP phones are configured for mulaw.
+
+Building
+---------
+
+Asterisk must be patched so that the core understand Codec 2 frames.
+
+1/ First install Codec 2:
+
+    david@cool:~$ svn co https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev codec2-dev
+    david@cool:~/codec2-dev$ cd codec2-dev
+    david@cool:~/codec2-dev$ ./configure && make && sudo make install
+    david@bear:~/codec2-dev$ sudo ldconfig -v
+    david@cool:~/codec2-dev$ cd ~
+
+2/ Then build Asterisk with Codec 2 support:
+
+    david@cool:~$ tar xvzf asterisk-1.8.9.0.tar.gz
+    david@cool:~/asterisk-1.8.9.0$ patch -p4 < ~/codec2-dev/asterisk/asterisk-codec2.patch
+    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec_codec2.c .
+    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h ./codecs
+    david@cool:~/asterisk-1.8.9.0$ ./configure && make ASTLDFLAGS=-lcodec2
+    david@cool:~/asterisk-1.8.9.0$ sudo make install
+    david@cool:~/asterisk-1.8.9.0$ sudo make samples
+
+3/ Add this to the end of sip.conf on Asterisk A:
+
+    [6013]
+    type=friend
+    context=default
+    host=dynamic
+    user=6013
+    secret=6013
+    canreinvite=no
+    callerid=6013
+    disallow=all
+    allow=ulaw
+
+    [potato]
+    type=peer
+    username=potato
+    fromuser=potato
+    secret=password
+    context=default
+    disallow=all
+    dtmfmode=rfc2833
+    callerid=server
+    canreinvite=no
+    host=cool
+    allow=codec2
+
+3/ Add this to the end of sip.conf on Asterisk B:
+
+    [6014]
+    type=friend
+    context=default
+    host=dynamic
+    user=6014
+    secret=6014
+    canreinvite=no
+    callerid=6014
+    disallow=all
+    allow=ulaw
+
+    [potato]
+    type=peer
+    username=potato
+    fromuser=potato
+    secret=password
+    context=default
+    disallow=all
+    dtmfmode=rfc2833
+    callerid=server
+    canreinvite=no
+    host=bear
+    allow=codec2
+
+4/ Here is the [default] section of extensions.conf on Asterisk B:
+
+    [default]
+
+    exten => 6013,1,Dial(SIP/potato/6013)
+    ;
+    ; By default we include the demo.  In a production system, you
+    ; probably don't want to have the demo there.
+    ;
+    ;include => demo
+
+5/ After booting see if the codec2_codec2.so module is loaded with "core show translate"
+
+6/ To make a test call dial 6013 on the SIP phone connected to Asterisk B
+
+7/ If codec_codec2.so won't load and you see "can't find codec2_create" try:
+
+    david@cool:~/asterisk-1.8.9.0$ touch codecs/codec_codec2.c
+    david@cool:~/asterisk-1.8.9.0$ make ASTLDFLAGS=-lcodec2
+    david@cool:~/asterisk-1.8.9.0$ sudo cp codecs/codec_codec2.so /usr/lib/asterisk/modules
+    david@cool:~/asterisk-1.8.9.0$ sudo asterisk -vvvcn
+

libs/libcodec2/asterisk/asterisk-codec2.patch

+--- /home/david/asterisk-1.8.9.0-orig/include/asterisk/frame.h	2011-12-23 05:08:46.000000000 +1030
++++ /home/david/asterisk-1.8.9.0-codec2/include/asterisk/frame.h	2012-03-27 13:16:55.623452431 +1030
+@@ -299,6 +299,7 @@
+ #define AST_FORMAT_G719	      (1ULL << 32)
+ /*! SpeeX Wideband (16kHz) Free Compression */
+ #define AST_FORMAT_SPEEX16    (1ULL << 33)
++#define AST_FORMAT_CODEC2    (1ULL << 34)
+ /*! Raw mu-law data (G.711) */
+ #define AST_FORMAT_TESTLAW    (1ULL << 47)
+ /*! Reserved bit - do not use */
+--- /home/david/asterisk-1.8.9.0-orig/main/frame.c	2010-06-18 02:53:43.000000000 +0930
++++ /home/david/asterisk-1.8.9.0-codec2/main/frame.c	2012-03-28 15:16:16.975581316 +1030
+@@ -102,6 +102,7 @@
+ 	{ AST_FORMAT_ADPCM, "adpcm" , 8000, "ADPCM", 40, 10, 300, 10, 20 },                                    /*!< codec_adpcm.c */
+ 	{ AST_FORMAT_SLINEAR, "slin", 8000, "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE }, /*!< Signed linear */
+ 	{ AST_FORMAT_LPC10, "lpc10", 8000, "LPC10", 7, 20, 20, 20, 20 },                                       /*!< codec_lpc10.c */ 
++	{ AST_FORMAT_CODEC2, "codec2", 8000, "Codec 2", 7, 20, 20, 20, 20 },                                       /*!< codec_codec2.c */ 
+ 	{ AST_FORMAT_G729A, "g729", 8000, "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 },             /*!< Binary commercial distribution */
+ 	{ AST_FORMAT_SPEEX, "speex", 8000, "SpeeX", 10, 10, 60, 10, 20 },                                      /*!< codec_speex.c */
+ 	{ AST_FORMAT_SPEEX16, "speex16", 16000, "SpeeX 16khz", 10, 10, 60, 10, 20 },                          /*!< codec_speex.c */
+@@ -1475,6 +1476,9 @@
+ 		samples = 22 * 8;
+ 		samples += (((char *)(f->data.ptr))[7] & 0x1) * 8;
+ 		break;
++	case AST_FORMAT_CODEC2:
++		samples = 160 * (f->datalen / 7);
++		break;
+ 	case AST_FORMAT_ULAW:
+ 	case AST_FORMAT_ALAW:
+ 	case AST_FORMAT_TESTLAW:
+@@ -1519,6 +1523,9 @@
+ 	case AST_FORMAT_GSM:
+ 		len = (samples / 160) * 33;
+ 		break;
++	case AST_FORMAT_CODEC2:
++		len = (samples / 160) * 7;
++		break;
+ 	case AST_FORMAT_G729A:
+ 		len = samples / 8;
+ 		break;
+--- /home/david/asterisk-1.8.9.0-orig/main/channel.c	2011-12-17 10:21:13.000000000 +1030
++++ /home/david/asterisk-1.8.9.0-codec2/main/channel.c	2012-03-28 15:05:22.395293391 +1030
+@@ -1075,6 +1075,7 @@
+ 		/*! Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough
+ 		    to use it */
+ 		AST_FORMAT_LPC10,
++		AST_FORMAT_CODEC2,
+ 		/*! G.729a is faster than 723 and slightly less expensive */
+ 		AST_FORMAT_G729A,
+ 		/*! Down to G.723.1 which is proprietary but at least designed for voice */
+--- /home/david/asterisk-1.8.9.0-orig/main/rtp_engine.c	2011-12-30 01:43:03.000000000 +1030
++++ /home/david/asterisk-1.8.9.0-codec2/main/rtp_engine.c	2012-03-28 16:42:02.880872891 +1030
+@@ -101,6 +101,7 @@
+ 	{{1, AST_FORMAT_SLINEAR}, "audio", "L16", 8000},
+ 	{{1, AST_FORMAT_SLINEAR16}, "audio", "L16", 16000},
+ 	{{1, AST_FORMAT_LPC10}, "audio", "LPC", 8000},
++	{{1, AST_FORMAT_CODEC2}, "audio", "CODEC2", 8000},
+ 	{{1, AST_FORMAT_G729A}, "audio", "G729", 8000},
+ 	{{1, AST_FORMAT_G729A}, "audio", "G729A", 8000},
+ 	{{1, AST_FORMAT_G729A}, "audio", "G.729", 8000},
+@@ -178,6 +179,7 @@
+ 	[117] = {1, AST_FORMAT_SPEEX16},
+ 	[118] = {1, AST_FORMAT_SLINEAR16}, /* 16 Khz signed linear */
+ 	[121] = {0, AST_RTP_CISCO_DTMF},   /* Must be type 121 */
++	[121] = {1, AST_FORMAT_CODEC2},
+ };
+ 
+ int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module)

libs/libcodec2/asterisk/codec_codec2.c

+/*
+ * Codec 2 module for Asterisk.
+ *
+ * Credit: codec_gsm.c used as a starting point.
+ *
+ * Copyright (C) 2012 Ed W and David Rowe
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Translate between signed linear and Codec 2
+ *
+ * \ingroup codecs
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/translate.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/utils.h"
+
+#include <codec2.h>
+
+#define BUFFER_SAMPLES	  8000
+#define CODEC2_SAMPLES    160
+#define	CODEC2_FRAME_LEN  7
+
+/* Sample frame data */
+
+#include "asterisk/slin.h"
+#include "ex_codec2.h"
+
+struct codec2_translator_pvt {	        /* both codec2tolin and codec2togsm */
+    struct CODEC2 *codec2;
+    short  buf[BUFFER_SAMPLES];	/* lintocodec2, temporary storage */
+};
+
+static int codec2_new(struct ast_trans_pvt *pvt)
+{
+    struct codec2_translator_pvt *tmp = pvt->pvt;
+
+    tmp->codec2 = codec2_create(CODEC2_MODE_2400);
+	
+    return 0;
+}
+
+/*! \brief decode and store in outbuf. */
+static int codec2tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+    struct codec2_translator_pvt *tmp = pvt->pvt;
+    int x;
+    int16_t *dst = pvt->outbuf.i16;
+    int flen = CODEC2_FRAME_LEN;
+
+    for (x=0; x < f->datalen; x += flen) {
+	unsigned char *src;
+	int len;
+	len = CODEC2_SAMPLES;
+	src = f->data.ptr + x;
+
+	codec2_decode(tmp->codec2, dst + pvt->samples, src);
+
+	pvt->samples += CODEC2_SAMPLES;
+	pvt->datalen += 2 * CODEC2_SAMPLES;
+    }
+    return 0;
+}
+
+/*! \brief store samples into working buffer for later decode */
+static int lintocodec2_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+	struct codec2_translator_pvt *tmp = pvt->pvt;
+
+	if (pvt->samples + f->samples > BUFFER_SAMPLES) {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	memcpy(tmp->buf + pvt->samples, f->data.ptr, f->datalen);
+	pvt->samples += f->samples;
+	return 0;
+}
+
+/*! \brief encode and produce a frame */
+static struct ast_frame *lintocodec2_frameout(struct ast_trans_pvt *pvt)
+{
+	struct codec2_translator_pvt *tmp = pvt->pvt;
+	int datalen = 0;
+	int samples = 0;
+
+	/* We can't work on anything less than a frame in size */
+	if (pvt->samples < CODEC2_SAMPLES)
+		return NULL;
+	while (pvt->samples >= CODEC2_SAMPLES) {
+	    /* Encode a frame of data */
+	    codec2_encode(tmp->codec2, (unsigned char*)(pvt->outbuf.c + datalen), tmp->buf + samples);
+	    datalen += CODEC2_FRAME_LEN;
+	    samples += CODEC2_SAMPLES;
+	    pvt->samples -= CODEC2_SAMPLES;
+	}
+
+	/* Move the data at the end of the buffer to the front */
+	if (pvt->samples)
+		memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
+
+	return ast_trans_frameout(pvt, datalen, samples);
+}
+
+static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
+{
+	struct codec2_translator_pvt *tmp = pvt->pvt;
+	if (tmp->codec2)
+		codec2_destroy(tmp->codec2);
+}
+
+static struct ast_translator codec2tolin = {
+	.name = "codec2tolin", 
+	.srcfmt = AST_FORMAT_CODEC2,
+	.dstfmt = AST_FORMAT_SLINEAR,
+	.newpvt = codec2_new,
+	.framein = codec2tolin_framein,
+	.destroy = codec2_destroy_stuff,
+	.sample = codec2_sample,
+	.buffer_samples = BUFFER_SAMPLES,
+	.buf_size = BUFFER_SAMPLES * 2,
+	.desc_size = sizeof (struct codec2_translator_pvt ),
+};
+
+static struct ast_translator lintocodec2 = {
+	.name = "lintocodec2", 
+	.srcfmt = AST_FORMAT_SLINEAR,
+	.dstfmt = AST_FORMAT_CODEC2,
+	.newpvt = codec2_new,
+	.framein = lintocodec2_framein,
+	.frameout = lintocodec2_frameout,
+	.destroy = codec2_destroy_stuff,
+	.sample = slin8_sample,
+	.desc_size = sizeof (struct codec2_translator_pvt ),
+	.buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1)/CODEC2_SAMPLES,
+};
+
+/*! \brief standard module glue */
+static int reload(void)
+{
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_translator(&lintocodec2);
+	if (!res)
+		res = ast_unregister_translator(&codec2tolin);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_translator(&codec2tolin);
+	if (!res) 
+		res=ast_register_translator(&lintocodec2);
+	else
+		ast_unregister_translator(&codec2tolin);
+	if (res) 
+		return AST_MODULE_LOAD_FAILURE;
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Codec 2 Coder/Decoder",
+		.load = load_module,
+		.unload = unload_module,
+		.reload = reload,
+	       );

libs/libcodec2/asterisk/ex_codec2.h

+/*! \file
+ * \brief 8-bit raw data
+ *
+ * Copyright (C) 2012, 2012 Ed W and David Rowe
+ *
+ * Distributed under the terms of the GNU General Public License
+ *
+ */
+
+static uint8_t ex_codec2[] = {
+    0x3e,0x06,0x4a,0xbb,0x9e,0x40,0xc0
+};
+
+static struct ast_frame *codec2_sample(void)
+{
+	static struct ast_frame f = {
+		.frametype = AST_FRAME_VOICE,
+		.subclass.codec = AST_FORMAT_CODEC2,
+		.datalen = sizeof(ex_codec2),
+		.samples = CODEC2_SAMPLES,
+		.mallocd = 0,
+		.offset = 0,
+		.src = __PRETTY_FUNCTION__,
+		.data.ptr = ex_codec2,
+	};
+
+	return &f;
+}

libs/libcodec2/asterisk/make_asterisk_patch.sh

+#!/bin/sh
+# Create patch for Codec 2 support inside Asterisk
+
+ORIG=~/asterisk-1.8.9.0-orig
+CODEC2=~/asterisk-1.8.9.0-codec2
+diff -ruN $ORIG/include/asterisk/frame.h $CODEC2/include/asterisk/frame.h > asterisk-codec2.patch
+diff -ruN $ORIG/main/frame.c $CODEC2/main/frame.c >> asterisk-codec2.patch
+diff -ruN $ORIG/main/channel.c $CODEC2/main/channel.c >> asterisk-codec2.patch
+diff -ruN $ORIG/main/rtp_engine.c $CODEC2/main/rtp_engine.c >> asterisk-codec2.patch
+
+

libs/libcodec2/fltk/Makefile

+# 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

libs/libcodec2/octave/README.txt

+README.txt
+For codec2/octave directory
+Created 24 June 2012 by David Rowe
+
+1/ To support some of the Octave scripts (in particular fdmdv) in this
+   directory the following Octave packages need to be installed:
+
+      control image miscellaneous optim signal specfun struct
+
+2/ Download these tar balls from:
+
+   http://octave.sourceforge.net/packages.php
+
+3/ Install each package from the Octave command line with:
+
+   octave:3> pkg install package_file_name.tar.gz
+

libs/libcodec2/octave/av_imp.m

+% av_imp.m
+% David Rowe Aug 2012
+% Averages the impulse response samples
+
+function imp = av_imp(imp_filename, period_in_secs, st, en)
+  f = fopen(imp_filename,"rb");
+  s = fread(f, Inf, "short")';
+
+  Fs = 8000;
+  n = period_in_secs * Fs;
+
+  [r c] = size(s);
+
+  imp = zeros(1,n);
+  for i=1:n:c-n
+    imp = imp + s(i:i+n-1);
+  endfor
+  
+  % user supplies start and end samples after viweing plot
+
+  if (nargin == 4)
+    imp = imp(st:en);
+  end
+
+  % normalise
+
+  imp /= sqrt(sum(imp .^ 2));
+
+  [h w] = freqz(imp, 1, 4000);
+
+  figure(1);
+  clf;
+  plot(imp);
+
+  figure(2);
+  clf;
+  subplot(211)
+  plot(10*log10(abs(h)))
+  subplot(212)
+  plot(angle(h))
+
+endfunction
+

libs/libcodec2/octave/cbphase.m

+% cbphase.m
+% David Rowe Aug 2012
+% Used to experiment with critical band phase perception and smoothing
+
+function cbphase
+
+  Wo = 100.0*pi/4000;
+  L = floor(pi/Wo);
+
+  A = zeros(1,L);
+  phi = zeros(1,L);
+
+  % three harmonics in this band
+
+  b = 4; a = b-1; c = b+1;
+
+  % set up phases and mags for 2nd order system (see phasesecord.m)
+   
+  wres = b*Wo;
+  phi(a) = 3*pi/4 + wres;
+  phi(b) = pi/2 + wres;
+  phi(c) = pi/4 + wres;
+
+  A(a) = 0.707;
+  A(b) = 1;
+  A(c) = 0.707;
+
+  % add linear component
+
+  phi(1) = pi;
+  phi(2:L) = phi(2:L) + (2:L)*phi(1);
+  phi = phi - 2*pi*(floor(phi/(2*pi)) + 0.5);
+
+  N = 16000;
+  Nplot = 250;
+  s = zeros(1,N);
+
+  for m=a:c
+    s_m = A(m)*cos(m*Wo*(0:(N-1)) + phi(m));
+    s = s + s_m;
+  endfor
+
+  figure(2);
+  clf;
+  subplot(211)
+  plot((1:L)*Wo*4000/pi, A,'+');
+  subplot(212)
+  plot((1:L)*Wo*4000/pi, phi,'+');
+
+  %v = A(a)*exp(j*phi(a)) + A(b)*exp(j*phi(b)) + A(c)*exp(j*phi(c));
+  %compass(v,"r")
+  %hold off;
+  
+  % est phi1
+
+  diff = phi(b) - phi(a)
+  sumi = sin(diff);
+  sumr = cos(diff);
+  diff = phi(c) - phi(b)
+  sumi += sin(diff);
+  sumr += cos(diff);
+  phi1_ = atan2(sumi, sumr)
+  s_v = cos(Wo*(0:(N-1)) + phi1_);
+
+  figure(1);
+  clf;
+  subplot(211)
+  plot(s(1:Nplot));
+  hold on;
+  plot(s_v(1:Nplot),"r");
+  hold off;
+
+  % build (hopefully) perceptually similar phase
+
+  phi_(a) = a*phi1_;
+  phi_(b) = b*phi1_;
+  phi_(c) = c*phi1_;
+
+  s_ = zeros(1,N);
+
+  for m=a:c
+    s_m = A(m)*cos(m*Wo*(0:(N-1)) + phi_(m));
+    s_ = s_ + s_m;
+  endfor
+ 
+  subplot(212)
+  plot(s_(1:Nplot));
+  
+  gain = 8000;
+  fs=fopen("orig_ph.raw","wb");
+  fwrite(fs,gain*s,"short");
+  fclose(fs);
+  fs=fopen("mod_ph.raw","wb");
+  fwrite(fs,gain*s_,"short");
+  fclose(fs);
+
+endfunction
+

libs/libcodec2/octave/codec2_demo.m

+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% codec2_demo.m
+
+% Designed as an educational tool to explain the operation of Codec 2
+% for conference and user group presentations on a projector.  An
+% alternative to static overhead slides.
+%
+% Derived from codec2-dev/octave/plamp.m
+%
+% usage:
+%   octave:1> plamp("../src/hts2a",40)
+%
+% Then press:
+%   c - to cycle through the wavform being displayed on the figure
+%   n - next frame
+%   b - back one frame
+%
+%   tip: hold down n or b to animate the display
+%
+% The text files used as input are generated using c2sim:
+%
+%   /codec2-dev/src$ c2sim ../raw/hts2a.raw --dump hts2a
+%
+% The Codec 2 README explains how to build c2sim with dump files
+% enabled.
+
+function codec2_demo(samname, f)
+  
+  sn_name = strcat(samname,"_sn.txt");
+  Sn = load(sn_name);
+
+  sw_name = strcat(samname,"_sw.txt");
+  Sw = load(sw_name);
+
+  model_name = strcat(samname,"_model.txt");
+  model = load(model_name);
+  
+  figure(1);
+
+  k = ' ';
+  wf = "Sn";
+  do 
+   
+    if strcmp(wf,"Sn")
+      clf;
+      s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+      plot(s);
+      axis([1 length(s) -20000 20000]);
+    end
+
+    if (strcmp(wf,"Sw"))
+      clf;
+      plot((0:255)*4000/256, Sw(f,:),";Sw;");
+    end
+  
+    if strcmp(wf,"SwAm")
+      Wo = model(f,1);
+      L = model(f,2);
+      Am = model(f,3:(L+2));
+      plot((0:255)*4000/256, Sw(f,:),";Sw;");
+      hold on;
+      plot((1:L)*Wo*4000/pi, 20*log10(Am),"+;Am;r");
+      axis([1 4000 -10 80]);
+      hold off;
+    end
+
+    if strcmp(wf,"Am")
+      Wo = model(f,1);
+      L = model(f,2);
+      Am = model(f,3:(L+2));
+      plot((1:L)*Wo*4000/pi, 20*log10(Am),"+;Am;r");
+      axis([1 4000 -10 80]);
+    end
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  w-cycle window  q-quit", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    end
+    if (k == 'b')
+      f = f - 1;
+    end
+    if (k == 'w') 
+      if strcmp(wf,"Sn")
+        next_wf = "Sw";
+      end
+      if strcmp(wf,"Sw")
+        next_wf = "SwAm";
+      end
+      if strcmp(wf,"SwAm")
+        next_wf = "Am";
+      end
+      if strcmp(wf,"Am")
+        next_wf = "Sn";
+      end
+      wf = next_wf;
+    end
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/cspec.m

+% cspec.m
+% David Rowe Aug 2012
+% Used to compare spectromgrams while experimenting with phase
+
+function cspec(s1,s2)
+  f1 = fopen(s1,"rb");
+  s1 = fread(f1,Inf,"short");
+  f2 = fopen(s2,"rb");
+  s2 = fread(f2,Inf,"short");
+
+  Fs = 8000;
+  spec_win = 512;
+
+  state = 's1';
+  do 
+    if strcmp(state,'s1')
+      spec(s1,Fs,spec_win);
+      %title(s1);
+    end
+    if strcmp(state,'s2')
+      spec(s2,Fs,spec_win);
+      %title(s2);
+    end
+    if strcmp(state,'diff')
+      spec(s1-s2,Fs,spec_win);
+      %title("difference");
+    end
+
+    printf("\rstate: %s  space-toggle d-diff q-quit", state);
+    fflush(stdout);
+    k = kbhit();
+    
+    if k == ' '
+      if strcmp(state,"diff")
+        next_state = 's1';
+      end
+      if strcmp(state,"s1")
+        next_state = 's2';
+      end
+      if strcmp(state,'s2')
+        next_state = 's1';
+      end
+    end
+
+    if k == 'd'
+      next_state = 'diff';
+    end
+
+    state = next_state;
+  until (k == 'q')
+
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/fdmdv_demod.m

+% fdmdv_demod.m
+%
+% Demodulator function for FDMDV modem (Octave version).  Requires
+% 8kHz sample rate raw files as input
+%
+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+
+function fdmdv_demod(rawfilename, nbits, pngname)
+
+  fdmdv; % include modem code
+
+  modulation = 'dqpsk';
+
+  fin = fopen(rawfilename, "rb");
+  gain = 1000;
+  frames = nbits/(Nc*Nb);
+
+  prev_rx_symbols = ones(Nc+1,1);
+  foff_phase = 1;
+
+  % BER stats
+
+  total_bit_errors = 0;
+  total_bits = 0;
+  bit_errors_log = [];
+  sync_log = [];
+  test_frame_sync_log = [];
+  test_frame_sync_state = 0;
+
+  % SNR states
+
+  sig_est = zeros(Nc+1,1);
+  noise_est = zeros(Nc+1,1);
+
+  % logs of various states for plotting
+
+  rx_symbols_log = [];
+  rx_timing_log = [];
+  foff_log = [];
+  rx_fdm_log = [];
+  snr_est_log = [];
+
+  % misc states
+
+  nin = M; % timing correction for sample rate differences
+  foff = 0;
+  track_log = [];
+  track = 0;
+  fest_state = 0;
+
+  % Main loop ----------------------------------------------------
+
+  for f=1:frames
+    
+    % obtain nin samples of the test input signal
+    
+    for i=1:nin
+      rx_fdm(i) = fread(fin, 1, "short")/gain;
+    end
+    
+    rx_fdm_log = [rx_fdm_log rx_fdm(1:nin)];
+
+    % frequency offset estimation and correction
+
+    [pilot prev_pilot pilot_lut_index prev_pilot_lut_index] = get_pilot(pilot_lut_index, prev_pilot_lut_index, nin);
+    [foff_coarse S1 S2] = rx_est_freq_offset(rx_fdm, pilot, prev_pilot, nin);
+    
+    if track == 0
+      foff  = foff_coarse;
+    end
+    foff_log = [ foff_log foff ];
+    foff_rect = exp(j*2*pi*foff/Fs);
+
+    for i=1:nin
+      foff_phase *= foff_rect';
+      rx_fdm(i) = rx_fdm(i)*foff_phase;
+    end
+
+    % baseband processing
+
+    rx_baseband = fdm_downconvert(rx_fdm, nin);
+    rx_filt = rx_filter(rx_baseband, nin);
+
+    [rx_symbols rx_timing] = rx_est_timing(rx_filt, rx_baseband, nin);
+    
+    rx_timing_log = [rx_timing_log rx_timing];
+    nin = M;
+    if rx_timing > 2*M/P
+       nin += M/P;
+    end
+    if rx_timing < 0;
+       nin -= M/P;
+    end
+
+    if strcmp(modulation,'dqpsk')
+      rx_symbols_log = [rx_symbols_log rx_symbols.*conj(prev_rx_symbols)*exp(j*pi/4)];
+    else
+      rx_symbols_log = [rx_symbols_log rx_symbols];
+    endif
+    [rx_bits sync f_err pd] = qpsk_to_bits(prev_rx_symbols, rx_symbols, modulation);
+    [sig_est noise_est] = snr_update(sig_est, noise_est, pd);
+    snr_est = calc_snr(sig_est, noise_est);
+    snr_est_log = [snr_est_log snr_est];
+    foff -= 0.5*f_err;
+    prev_rx_symbols = rx_symbols;
+    sync_log = [sync_log sync];
+
+    % freq est state machine
+
+    [track fest_state] = freq_state(sync, fest_state);
+    track_log = [track_log track];
+
+    % count bit errors if we find a test frame
+
+    [test_frame_sync bit_errors] = put_test_bits(rx_bits);
+    if (test_frame_sync == 1)
+      total_bit_errors = total_bit_errors + bit_errors;
+      total_bits = total_bits + Ntest_bits;
+      bit_errors_log = [bit_errors_log bit_errors/Ntest_bits];
+    else
+      bit_errors_log = [bit_errors_log 0];
+    end
+
+    % test frame sync state machine, just for more informative plots
+    
+    next_test_frame_sync_state = test_frame_sync_state;
+    if (test_frame_sync_state == 0)
+      if (test_frame_sync == 1)      
+        next_test_frame_sync_state = 1;
+	test_frame_count = 0;
+      end
+    end
+
+    if (test_frame_sync_state == 1)
+      % we only expect another test_frame_sync pulse every 4 symbols
+      test_frame_count++;
+      if (test_frame_count == 4)
+        test_frame_count = 0;
+        if ((test_frame_sync == 0))      
+          next_test_frame_sync_state = 0;
+        end
+      end
+    end
+    test_frame_sync_state = next_test_frame_sync_state;
+    test_frame_sync_log = [test_frame_sync_log test_frame_sync_state];
+
+  end
+  
+  % ---------------------------------------------------------------------
+  % Print Stats
+  % ---------------------------------------------------------------------
+
+  ber = total_bit_errors / total_bits;
+
+  printf("%d bits  %d errors  BER: %1.4f\n",total_bits, total_bit_errors, ber);
+
+  % ---------------------------------------------------------------------
+  % Plots
+  % ---------------------------------------------------------------------
+
+  xt = (1:frames)/Rs;
+  secs = frames/Rs;
+
+  figure(1)
+  clf;
+  [n m] = size(rx_symbols_log);
+  plot(real(rx_symbols_log(1:Nc+1,15:m)),imag(rx_symbols_log(1:Nc+1,15:m)),'+')
+  axis([-2 2 -2 2]);
+  title('Scatter Diagram');
+
+  figure(2)
+  clf;
+  subplot(211)
+  plot(xt, rx_timing_log)
+  title('timing offset (samples)');
+  subplot(212)
+  plot(xt, foff_log, '-;freq offset;')
+  hold on;
+  plot(xt, track_log*75, 'r;course-fine;');
+  hold off;
+  title('Freq offset (Hz)');
+  grid
+
+  figure(3)
+  clf;
+  subplot(211)
+  [a b] = size(rx_fdm_log);
+  xt1 = (1:b)/Fs;
+  plot(xt1, rx_fdm_log);
+  title('Rx FDM Signal');
+  subplot(212)
+  spec(rx_fdm_log,8000);
+  title('FDM Rx Spectrogram');
+
+  figure(4)
+  clf;
+  subplot(311)
+  stem(xt, sync_log)
+  axis([0 secs 0 1.5]);
+  title('BPSK Sync')
+  subplot(312)
+  stem(xt, bit_errors_log);
+  title('Bit Errors for test frames')
+  subplot(313)
+  plot(xt, test_frame_sync_log);
+  axis([0 secs 0 1.5]);
+  title('Test Frame Sync')
+
+  figure(5)
+  clf;
+  plot(xt, snr_est_log);
+  title('SNR Estimates')
+ 
+endfunction

libs/libcodec2/octave/fdmdv_demod_c.m

+% fdmdv_demod_c.m
+%
+% Plots Octave dump file information from C FDMDV demodulator program,
+% to give a similar set of plots to fdmdv_demod.m.  Useful for off
+% line analysis of demod performance.
+%
+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+
+function fdmdv_demod_c(dumpfilename, bits)
+
+  fdmdv; % include modem code
+
+  frames = bits/(Nc*Nb);
+
+  load(dumpfilename);
+
+  % BER stats
+
+  total_bit_errors = 0;
+  total_bits = 0;
+  bit_errors_log = [];
+  sync_log = [];
+  test_frame_sync_log = [];
+  test_frame_sync_state = 0;
+
+  % Run thru received bits to look for test pattern
+
+  bits_per_frame = Nc*Nb;
+
+  for f=1:frames
+
+    rx_bits = rx_bits_log_c((f-1)*bits_per_frame+1:f*bits_per_frame);
+
+    % count bit errors if we find a test frame
+
+    [test_frame_sync bit_errors] = put_test_bits(rx_bits);
+    if (test_frame_sync == 1)
+      total_bit_errors = total_bit_errors + bit_errors;
+      total_bits = total_bits + Ntest_bits;
+      bit_errors_log = [bit_errors_log bit_errors/Ntest_bits];
+    else
+      bit_errors_log = [bit_errors_log 0];
+    end
+
+    % test frame sync state machine, just for more informative plots
+    
+    next_test_frame_sync_state = test_frame_sync_state;
+    if (test_frame_sync_state == 0)
+      if (test_frame_sync == 1)      
+        next_test_frame_sync_state = 1;
+	test_frame_count = 0;
+      end
+    end
+
+    if (test_frame_sync_state == 1)
+      % we only expect another test_frame_sync pulse every 4 symbols
+      test_frame_count++;
+      if (test_frame_count == 4)
+        test_frame_count = 0;
+        if ((test_frame_sync == 0))      
+          next_test_frame_sync_state = 0;
+        end
+      end
+    end
+    test_frame_sync_state = next_test_frame_sync_state;
+    test_frame_sync_log = [test_frame_sync_log test_frame_sync_state];
+  end
+
+  % ---------------------------------------------------------------------
+  % Plots
+  % ---------------------------------------------------------------------
+
+  xt = (1:frames)/Rs;
+  secs = frames/Rs;
+
+  figure(1)
+  clf;
+  plot(real(rx_symbols_log_c(1:Nc+1,15:frames)),imag(rx_symbols_log_c(1:Nc+1,15:frames)),'+')
+  axis([-2 2 -2 2]);
+  title('Scatter Diagram');
+
+  figure(2)
+  clf;
+  subplot(211)
+  plot(xt, rx_timing_log_c(1:frames))
+  title('timing offset (samples)');
+  subplot(212)
+  plot(xt, foff_log_c(1:frames), '-;freq offset;')
+  hold on;
+  plot(xt, coarse_fine_log_c(1:frames)*75, 'r;course-fine;');
+  hold off;
+  title('Freq offset (Hz)');
+  grid
+
+  figure(3)
+  clf;
+  subplot(211)
+  b = M*frames;
+  xt1 = (1:b)/Fs;
+  plot(xt1, rx_fdm_log_c(1:b));
+  title('Rx FDM Signal');
+  subplot(212)
+  spec(rx_fdm_log_c(1:b),8000);
+  title('FDM Rx Spectrogram');
+
+  figure(4)
+  clf;
+  subplot(311)
+  stem(xt, sync_bit_log_c(1:frames))
+  axis([0 secs 0 1.5]);
+  title('BPSK Sync')
+  subplot(312)
+  stem(xt, bit_errors_log);
+  title('Bit Errors for test frames')
+  subplot(313)
+  plot(xt, test_frame_sync_log);
+  axis([0 secs 0 1.5]);
+  title('Test Frame Sync')
+
+  figure(5)
+  clf;
+  plot(xt, snr_est_log_c(1:frames));
+  title('SNR Estimates')
+
+endfunction

libs/libcodec2/octave/fdmdv_mod.m

+% fdmdv_mod.m
+%
+% Modulator function for FDMDV modem, uses test frames as input and
+% outputs a raw file of 16 bit shorts at a sample rate of 8 kHz.
+%
+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+
+function tx_fdm = fdmdv_mod(rawfilename, nbits)
+
+  fdmdv; % include modem code
+
+  frames = floor(nbits/(Nc*Nb))
+  tx_fdm = [];
+  gain = 1000; % Scale up to 16 bit shorts
+  prev_tx_symbols = ones(Nc+1,1);
+
+  for i=1:frames
+    tx_bits = get_test_bits(Nc*Nb);
+    tx_symbols = bits_to_qpsk(prev_tx_symbols, tx_bits,'dqpsk');
+    prev_tx_symbols = tx_symbols;
+    tx_baseband = tx_filter(tx_symbols);
+    tx_fdm = [tx_fdm real(fdm_upconvert(tx_baseband))];
+  end
+
+  tx_fdm *= gain;
+  fout = fopen(rawfilename,"wb");
+  fwrite(fout, tx_fdm, "short");
+  fclose(fout);
+endfunction

libs/libcodec2/octave/fdmdv_ut.m

+% fdmdv_ut.m
+%
+% Unit Test program for FDMDV modem.  Useful for general development as it has
+% both tx and rx sides, and basic AWGN channel simulation.
+%
+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+
+fdmdv;               % load modem code
+ 
+% Simulation Parameters --------------------------------------
+
+frames = 25;
+EbNo_dB = 7.3;
+Foff_hz = 0;
+modulation = 'dqpsk';
+hpa_clip = 150;
+
+% ------------------------------------------------------------
+
+tx_filt = zeros(Nc,M);
+rx_symbols_log = [];
+rx_phase_log = 0;
+rx_timing_log = 0;
+tx_pwr = 0;
+noise_pwr = 0;
+rx_fdm_log = [];
+rx_baseband_log = [];
+rx_bits_offset = zeros(Nc*Nb*2);
+prev_tx_symbols = ones(Nc+1,1);
+prev_rx_symbols = ones(Nc+1,1);
+ferr = 0;
+foff = 0;
+foff_log = [];
+tx_baseband_log = [];
+tx_fdm_log = [];
+
+% BER stats
+
+total_bit_errors = 0;
+total_bits = 0;
+bit_errors_log = [];
+sync_log = [];
+test_frame_sync_log = [];
+test_frame_sync_state = 0;
+
+% SNR estimation states
+
+sig_est = zeros(Nc+1,1);
+noise_est = zeros(Nc+1,1);
+
+% fixed delay simuation
+
+Ndelay = M+20;
+rx_fdm_delay = zeros(Ndelay,1);
+
+% ---------------------------------------------------------------------
+% Eb/No calculations.  We need to work out Eb/No for each FDM carrier.
+% Total power is sum of power in all FDM carriers
+% ---------------------------------------------------------------------
+
+C = 1; % power of each FDM carrier (energy/sample).  Total Carrier power should = Nc*C = Nc
+N = 1; % total noise power (energy/sample) of noise source across entire bandwidth
+
+% Eb  = Carrier power * symbol time / (bits/symbol)
+%     = C *(1/Rs) / 2
+Eb_dB = 10*log10(C) - 10*log10(Rs) - 10*log10(2);
+
+No_dBHz = Eb_dB - EbNo_dB;
+
+% Noise power = Noise spectral density * bandwidth
+% Noise power = Noise spectral density * Fs/2 for real signals
+N_dB = No_dBHz + 10*log10(Fs/2);
+Ngain_dB = N_dB - 10*log10(N);
+Ngain = 10^(Ngain_dB/20);
+
+% C/No = Carrier Power/noise spectral density
+%      = power per carrier*number of carriers / noise spectral density
+CNo_dB = 10*log10(C)  + 10*log10(Nc) - No_dBHz;
+
+% SNR in equivalent 3000 Hz SSB channel
+
+B = 3000;
+SNR = CNo_dB - 10*log10(B);
+
+% freq offset simulation states
+
+phase_offset = 1;
+freq_offset = exp(j*2*pi*Foff_hz/Fs);
+foff_phase = 1;
+t = 0;
+foff = 0;
+fest_state = 0;
+track = 0;
+track_log = [];
+
+
+% ---------------------------------------------------------------------
+% Main loop 
+% ---------------------------------------------------------------------
+
+for f=1:frames
+
+  % -------------------
+  % Modulator
+  % -------------------
+
+  tx_bits = get_test_bits(Nc*Nb);
+  tx_symbols = bits_to_qpsk(prev_tx_symbols, tx_bits, modulation);
+  prev_tx_symbols = tx_symbols;
+  tx_baseband = tx_filter(tx_symbols);
+  tx_baseband_log = [tx_baseband_log tx_baseband];
+  tx_fdm = fdm_upconvert(tx_baseband);
+  tx_pwr = 0.9*tx_pwr + 0.1*real(tx_fdm)*real(tx_fdm)'/(M);
+
+  % -------------------
+  % Channel simulation
+  % -------------------
+
+  % frequency offset
+
+  %Foff_hz += 1/Rs;
+  Foff = Foff_hz;
+  for i=1:M
+    % Time varying freq offset
+    %Foff = Foff_hz + 100*sin(t*2*pi/(300*Fs));
+    %t++;
+    freq_offset = exp(j*2*pi*Foff/Fs);
+    phase_offset *= freq_offset;
+    tx_fdm(i) = phase_offset*tx_fdm(i);
+  end
+
+  tx_fdm = real(tx_fdm);
+
+  % HPA non-linearity
+
+  tx_fdm(find(abs(tx_fdm) > hpa_clip)) = hpa_clip;
+  tx_fdm_log = [tx_fdm_log tx_fdm];
+
+  rx_fdm = tx_fdm;
+
+  % AWGN noise
+
+  noise = Ngain*randn(1,M);
+  noise_pwr = 0.9*noise_pwr + 0.1*noise*noise'/M;
+  rx_fdm += noise;
+  rx_fdm_log = [rx_fdm_log rx_fdm];
+
+  % Delay
+
+  rx_fdm_delay(1:Ndelay-M) = rx_fdm_delay(M+1:Ndelay);
+  rx_fdm_delay(Ndelay-M+1:Ndelay) = rx_fdm;
+  %rx_fdm_delay = rx_fdm;
+
+  % -------------------
+  % Demodulator
+  % -------------------
+
+  % frequency offset estimation and correction, need to call rx_est_freq_offset even in track
+  % mode to keep states updated
+
+  [pilot prev_pilot pilot_lut_index prev_pilot_lut_index] = get_pilot(pilot_lut_index, prev_pilot_lut_index, M);
+  [foff_course S1 S2] = rx_est_freq_offset(rx_fdm_delay, pilot, prev_pilot, M);
+  if track == 0
+    foff = foff_course;
+  end
+  foff_log = [ foff_log foff ];
+  foff_rect = exp(j*2*pi*foff/Fs);
+
+  for i=1:M
+    foff_phase *= foff_rect';
+    rx_fdm_delay(i) = rx_fdm_delay(i)*foff_phase;
+  end
+
+  % baseband processing
+
+  rx_baseband = fdm_downconvert(rx_fdm_delay(1:M), M);
+  rx_baseband_log = [rx_baseband_log rx_baseband];
+  rx_filt = rx_filter(rx_baseband, M);
+
+  [rx_symbols rx_timing] = rx_est_timing(rx_filt, rx_baseband, M);
+  rx_timing_log = [rx_timing_log rx_timing];
+
+  %rx_phase = rx_est_phase(rx_symbols);
+  %rx_phase_log = [rx_phase_log rx_phase];
+  %rx_symbols = rx_symbols*exp(j*rx_phase);
+
+  [rx_bits sync foff_fine pd] = qpsk_to_bits(prev_rx_symbols, rx_symbols, modulation);
+  if strcmp(modulation,'dqpsk')
+    %rx_symbols_log = [rx_symbols_log rx_symbols.*conj(prev_rx_symbols)*exp(j*pi/4)];
+    rx_symbols_log = [rx_symbols_log pd];
+  else
+    rx_symbols_log = [rx_symbols_log rx_symbols];
+  endif
+  foff -= 0.5*ferr;
+  prev_rx_symbols = rx_symbols;
+  sync_log = [sync_log sync];
+  
+  % freq est state machine
+
+  [track fest_state] = freq_state(sync, fest_state);
+  track_log = [track_log track];
+
+  % Update SNR est
+
+  [sig_est noise_est] = snr_update(sig_est, noise_est, pd);
+
+  % count bit errors if we find a test frame
+  % Allow 15 frames for filter memories to fill and time est to settle
+
+  [test_frame_sync bit_errors] = put_test_bits(rx_bits);
+  if test_frame_sync == 1
+    total_bit_errors = total_bit_errors + bit_errors;
+    total_bits = total_bits + Ntest_bits;
+    bit_errors_log = [bit_errors_log bit_errors];
+    else
+      bit_errors_log = [bit_errors_log 0];
+  end
+ 
+  % test frame sync state machine, just for more informative plots
+    
+  next_test_frame_sync_state = test_frame_sync_state;
+  if (test_frame_sync_state == 0)
+    if (test_frame_sync == 1)      
+      next_test_frame_sync_state = 1;
+      test_frame_count = 0;
+    end
+  end
+
+  if (test_frame_sync_state == 1)
+    % we only expect another test_frame_sync pulse every 4 symbols
+    test_frame_count++;
+    if (test_frame_count == 4)
+      test_frame_count = 0;
+      if ((test_frame_sync == 0))      
+        next_test_frame_sync_state = 0;
+      end
+    end
+  end
+  test_frame_sync_state = next_test_frame_sync_state;
+  test_frame_sync_log = [test_frame_sync_log test_frame_sync_state];
+end
+
+% ---------------------------------------------------------------------
+% Print Stats
+% ---------------------------------------------------------------------
+
+ber = total_bit_errors / total_bits;
+
+% Peak to Average Power Ratio calcs from http://www.dsplog.com
+
+papr = max(tx_fdm_log.*conj(tx_fdm_log)) / mean(tx_fdm_log.*conj(tx_fdm_log));
+papr_dB = 10*log10(papr);
+
+% Note Eb/No set point is for Nc data carriers only, exclduing pilot.
+% This is convenient for testing BER versus Eb/No.  Measured Eb/No
+% includes power of pilot.  Similar for SNR, first number is SNR excluding
+% pilot pwr for Eb/No set point, 2nd value is measured SNR which will be a little
+% higher as pilot power is included.
+
+printf("Eb/No (meas): %2.2f (%2.2f) dB\n", EbNo_dB, 10*log10(0.25*tx_pwr*Fs/(Rs*Nc*noise_pwr)));
+printf("bits........: %d\n", total_bits);
+printf("errors......: %d\n", total_bit_errors);
+printf("BER.........: %1.4f\n",  ber);
+printf("PAPR........: %1.2f dB\n", papr_dB);
+printf("SNR...(meas): %2.2f (%2.2f) dB\n", SNR, calc_snr(sig_est, noise_est));
+
+% ---------------------------------------------------------------------
+% Plots
+% ---------------------------------------------------------------------
+
+figure(1)
+clf;
+[n m] = size(rx_symbols_log);
+plot(real(rx_symbols_log(1:Nc+1,15:m)),imag(rx_symbols_log(1:Nc+1,15:m)),'+')
+axis([-3 3 -3 3]);
+title('Scatter Diagram');
+
+figure(2)
+clf;
+subplot(211)
+plot(rx_timing_log)
+title('timing offset (samples)');
+subplot(212)
+plot(foff_log, '-;freq offset;')
+hold on;
+plot(track_log*75, 'r;course-fine;');
+hold off;
+title('Freq offset (Hz)');
+
+figure(3)
+clf;
+subplot(211)
+plot(real(tx_fdm_log));
+title('FDM Tx Signal');
+subplot(212)
+Nfft=Fs;
+S=fft(rx_fdm_log,Nfft);
+SdB=20*log10(abs(S));
+plot(SdB(1:Fs/4))
+title('FDM Rx Spectrum');
+
+figure(4)
+clf;
+subplot(311)
+stem(sync_log)
+axis([0 frames 0 1.5]);
+title('BPSK Sync')
+subplot(312)
+stem(bit_errors_log);
+title('Bit Errors for test frames')
+subplot(313)
+plot(test_frame_sync_log);
+axis([0 frames 0 1.5]);
+title('Test Frame Sync')
+

libs/libcodec2/octave/gen_rn_coeffs.m

+% gen_rn_coeffs.m
+% David Rowe 13 april 2012
+%
+% Generate root raised cosine (Root Nyquist) filter coefficients
+% thanks http://www.dsplog.com/db-install/wp-content/uploads/2008/05/raised_cosine_filter.m
+
+function coeffs = gen_rn_coeffs(alpha, T, Rs, Nsym, M)
+
+  Ts = 1/Rs;
+
+  n = -Nsym*Ts/2:T:Nsym*Ts/2;
+  Nfilter = Nsym*M;
+  Nfiltertiming = M+Nfilter+M;
+
+  sincNum = sin(pi*n/Ts); % numerator of the sinc function
+  sincDen = (pi*n/Ts);    % denominator of the sinc function
+  sincDenZero = find(abs(sincDen) < 10^-10);
+  sincOp = sincNum./sincDen;
+  sincOp(sincDenZero) = 1; % sin(pix/(pix) =1 for x =0
+
+  cosNum = cos(alpha*pi*n/Ts);
+  cosDen = (1-(2*alpha*n/Ts).^2);
+  cosDenZero = find(abs(cosDen)<10^-10);
+  cosOp = cosNum./cosDen;
+  cosOp(cosDenZero) = pi/4;
+  gt_alpha5 = sincOp.*cosOp;
+  Nfft = 4096;
+  GF_alpha5 = fft(gt_alpha5,Nfft)/M;
+
+  % sqrt causes stop band to be amplified, this hack pushes it down again
+
+  for i=1:Nfft
+    if (abs(GF_alpha5(i)) < 0.02)
+      GF_alpha5(i) *= 0.001;
+    endif
+  end
+  GF_alpha5_root = sqrt(abs(GF_alpha5)) .* exp(j*angle(GF_alpha5));
+  ifft_GF_alpha5_root = ifft(GF_alpha5_root);
+  coeffs = real((ifft_GF_alpha5_root(1:Nfilter)));
+endfunction

libs/libcodec2/octave/hp_filt.m

+% hp_filt.m
+% David Rowe 20 Feb 2012
+
+function hp_filt(in_file, out_file)
+  fin = fopen(in_file,"rb");
+  s = fread(fin,Inf,"short");
+  b = fir1(256, 300/4000, "high");
+  freqz(b);
+  s_hpf = filter(b,1,s);
+  fout = fopen(out_file,"wb");
+  fwrite(fout, s_hpf, "short");
+endfunction

libs/libcodec2/octave/lpcpf.m

+% lpcpf.m
+% David Rowe Aug 27 2012
+% Experiments with LPC post filtering
+
+function lpcpf(ak_filename, f)
+  aks = load(ak_filename);
+	 
+  ak = aks(f,:);
+  [tmp p] = size(ak);
+  p -= 1;
+
+  A = freqz(1,ak, 4000);	 
+  AdB = 20*log10(abs(A));
+
+  gamma = 0.5;
+  gammas = gamma .^ (0:p);
+  W = freqz(ak .* gammas,1, 4000);
+  WdB = 20*log10(abs(W));
+
+  beta = 0.2;
+  R = abs(freqz(ak .* gammas, ak, 4000));
+  %P = (R/max(R)) .^ beta;
+  P = R .^ beta;
+  AP = abs(A) .* P;
+
+  eA = sum(abs(A) .^ 2);
+  eAP = sum(AP .^ 2);
+  gain = sqrt(eA/eAP)
+  AP *= gain;
+
+  PdB = 20*log10(P);
+
+  APdB = 20*log10(AP);
+  10*log10(sum(AP .^ 2))/10*log10(sum(abs(A) .^ 2))
+
+  figure(1);
+  clf;
+  plot(AdB);
+  hold on;
+  plot(WdB,'g');
+  plot(PdB,'r');
+  plot(APdB,'b.');
+  hold off;
+
+endfunction
+

libs/libcodec2/octave/lspwarp.m

+% lspwarp.m
+% David Rowe Sep 2012
+%
+% Experimenting with non-linear LSP frequency axis for LSP quantisation
+% Plots a scaled mel axis.
+
+1;
+
+function mel = freq2mel(f)
+  mel = 70*log10(1 + f/700);
+endfunction
+
+function freq = mel2freq(m)
+  freq = 700*(10 ^ (m/70) - 1);
+endfunction
+
+x = []; y = [];
+
+for freq = 100:25:4000
+  mel = freq2mel(freq);
+  x = [x freq];
+  y = [y mel];
+end
+
+plot(x,y)
+grid
+
+mel_start = floor(freq2mel(100));
+mel_end = floor(freq2mel(4000));
+
+x = []; y = [];
+for mel=mel_start:mel_end
+  freq = mel2freq(mel);
+  x = [x freq];
+  y = [y mel];
+end
+
+hold on;
+plot(x,y, '+')
+hold off;

libs/libcodec2/octave/phasesecord.m

+% phasesecord.m
+% David Rowe Aug 2012
+% Used to experiment with aproximations of phase of 2nd order systems
+
+function phasesecord(w,beta)
+
+  a = [1 -2*cos(w)*beta beta*beta];
+  b = 1;
+
+  [h w1] = freqz(b,a);
+
+  figure(1)
+  subplot(211)
+  plot(abs(h))
+  subplot(212)
+  plot(angle(h))
+
+  % for beta close to 1, we approximate 3 dB points as 1-beta above
+  % and below the resonance freq. Note this fails if w=0 as there is a
+  % double pole.  Lets sample the freq response at the 3dB points and
+  % w:
+
+  ws = [w-(1-beta) w w+(1-beta)];
+  [h w1] = freqz(b,a,ws);
+
+  % gain as a fraction of max, should be 3dB. Within 1.3 dB or for w > pi/8,
+  % gets innacurate near w=0 due to 2nd pole
+
+  printf("mag measured...:"); printf("% 4.3f ", abs(h)/max(abs(h)));
+
+  % measured angle, 45 deg from angle at w
+
+  printf("\nangle measured.: "); printf("% 5.3f ", angle(h));
+
+  % Our estimate of angle, (pi+w) is phase at resonance, at lower 3dB
+  % phase is pi/4 ahead, at upper 3B pi/4 behind.  -pi/2 is contribution of
+  % other pole at at -w to phase
+
+  ph_lower = (pi+w) + pi/4 - pi/2;
+  ph_res   =(pi+w) - pi/2;
+  ph_upper = (pi+w) - pi/4 - pi/2;
+  ph_ests = [ph_lower ph_res ph_upper];
+  ph_ests = ph_ests - 2*pi*(floor(ph_ests/(2*pi)) + 0.5);
+  printf("\nangle estimated:"); printf("% 5.3f ", ph_ests);
+  printf("\n");
+endfunction
+

libs/libcodec2/octave/pllpcpf.m

+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot amplitude modelling information from dump files to test and develop
+% LPC post filter.
+
+function pllpcpf(samname, f)
+  
+  % switch some stuff off to unclutter display
+
+  plot_Am = 0;
+  plot_Amq = 0;
+  plot_err = 0;
+  plot_lsp = 0;
+  plot_snr = 0;
+  plot_vsnr = 0;
+  plot_sw = 0;
+  plot_pw = 1;
+  plot_pwb = 1;
+  plot_rw = 1;
+
+  sn_name = strcat(samname,"_sn.txt");
+  Sn = load(sn_name);
+
+  sw_name = strcat(samname,"_sw.txt");
+  Sw = load(sw_name);
+
+  sw__name = strcat(samname,"_sw_.txt");
+  if (file_in_path(".",sw__name))
+    Sw_ = load(sw__name);
+  endif
+
+  model_name = strcat(samname,"_model.txt");
+  model = load(model_name);
+
+  modelq_name = strcat(samname,"_qmodel.txt");
+  if (file_in_path(".",modelq_name))
+    modelq = load(modelq_name);
+  endif
+
+  % Pw (LPC synth filter spectrum) before post filter
+
+  pwb_name = strcat(samname,"_pwb.txt");
+  if (file_in_path(".",pwb_name))
+    Pwb = load(pwb_name);
+  endif
+
+  % Rw (Post filter spectrum)
+
+  rw_name = strcat(samname,"_rw.txt");
+  if (file_in_path(".",rw_name))
+    Rw = load(rw_name);
+  endif
+
+  % Pw (LPC synth filter spectrum) after post filter
+
+  pw_name = strcat(samname,"_pw.txt");
+  if (file_in_path(".",pw_name))
+    Pw = load(pw_name);
+  endif
+
+
+  Ew_on = 1;
+  k = ' ';
+  do 
+    figure(1);
+    clf;
+    s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+    size(s);
+    plot(s);
+    axis([1 length(s) -20000 20000]);
+
+    figure(2);
+    clf;
+    Wo = model(f,1);
+    L = model(f,2);
+    Am = model(f,3:(L+2));
+    if plot_Am
+      plot((1:L)*Wo*4000/pi, 20*log10(Am),";Am;r");
+    end
+    axis([1 4000 -10 80]);
+    hold on;
+    if plot_sw
+      plot((0:255)*4000/256, Sw(f,:),";Sw;");
+    end
+ 
+    if (file_in_path(".",modelq_name))
+
+      Amq = modelq(f,3:(L+2));
+      if plot_Amq
+        plot((1:L)*Wo*4000/pi, 20*log10(Amq),";Amq;g" );
+      end
+
+      if (file_in_path(".",pwb_name) && plot_pwb)
+        plot((0:255)*4000/256, 10*log10(Pwb(f,:)),";Pwb;r");
+      endif	
+
+      if (file_in_path(".",rw_name) && plot_rw)
+        plot((0:255)*4000/256, 10*log10(Rw(f,:)),";Rw;b");
+      endif	
+
+      if (file_in_path(".",pw_name) && plot_pw)
+        plot((0:255)*4000/256, 10*log10(Pw(f,:)),";Pw;g.");
+      endif	
+
+      signal = Am * Am';
+      noise = (Am-Amq) * (Am-Amq)'; 
+      snr1 = 10*log10(signal/noise);
+      Am_err_label = sprintf(";Am error SNR %4.2f dB;m",snr1);
+      if plot_err
+        plot((1:L)*Wo*4000/pi, 20*log10(Amq) - 20*log10(Am), Am_err_label);
+      end
+    endif
+
+
+    hold off;
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  p-png  q-quit", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+    % optional print to PNG
+
+    if (k == 'p')
+      figure(1);
+      pngname = sprintf("%s_%d_sn.png",samname,f);
+      print(pngname, '-dpng', "-S500,500")
+      pngname = sprintf("%s_%d_sn_large.png",samname,f);
+      print(pngname, '-dpng', "-S800,600")
+
+      figure(2);
+      pngname = sprintf("%s_%d_sw.png",samname,f);
+      print(pngname, '-dpng', "-S500,500")
+      pngname = sprintf("%s_%d_sw_large.png",samname,f);
+      print(pngname, '-dpng', "-S1200,800")
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/pllsp.m

+% Copyright David Rowe 2010
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plots a bunch of information related to LSP quantisation:
+%   - speech file 
+%   - LSPs before and after quantisation
+%   - SNR for each frame
+%
+% Note: there is a 160 sample (two frame delay) from the when a sample
+% enters the input buffer until it is at the centre of the analysis window
+
+function pllsp(rawfile, 
+	       dumpfile_prefix_lpc_only, 
+               dumpfile_prefix_lsp, 
+               start_f, end_f)
+  
+  fs=fopen(rawfile,"rb");
+  s=fread(fs,Inf,"short");
+
+  lpc_snr_name = strcat(dumpfile_prefix_lpc_only,"_lpc_snr.txt");
+  lpc10_snr = load(lpc_snr_name);
+  lpc_snr_name = strcat(dumpfile_prefix_lsp,"_lpc_snr.txt");
+  lsp_snr = load(lpc_snr_name);
+
+  lsp_name = strcat(dumpfile_prefix_lsp,"_lsp.txt");
+  lsps = load(lsp_name);
+  [m,n]=size(lsps);
+  lsp  = lsps(1:2:m,:);
+  lsp_ = lsps(2:2:m,:);
+
+  figure(1);
+  clf;
+  subplot(211);
+  sp = s((start_f-2)*80:(end_f-2)*80);
+  plot(sp);
+
+  subplot(212);
+  plot(lpc10_snr((start_f+1):end_f)-lsp_snr((start_f+1):end_f));
+
+  figure(2);
+  plot((4000/pi)*lsp((start_f+1):end_f,:));
+  hold on;
+  plot((4000/pi)*lsp_((start_f+1):end_f,:),'+-');
+  hold off;
+endfunction

libs/libcodec2/octave/pllspdt.m

+% pllspdt.m
+% Copyright David Rowe 2010
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Test script to plot differences in LSps between frames
+
+function pllspdt(rawfile,dumpfile_prefix_lsp,lspn, start_f, end_f)
+  
+  fs=fopen(rawfile,"rb");
+  s=fread(fs,Inf,"short");
+
+  lsp_name = strcat(dumpfile_prefix_lsp,"_lsp.txt");
+  lsps = load(lsp_name);
+  [m,n]=size(lsps);
+  lsp  = lsps(1:2:m,:);
+  lsp_ = lsps(2:2:m,:);
+  lspdt = lsp(2:m/2,:) - lsp(1:m/2-1,:);
+
+  figure(1);
+  clf;
+  sp = s((start_f-2)*80:(end_f-2)*80);
+  plot(sp);
+
+  figure(2);
+  plot((4000/pi)*lspdt((start_f+1):end_f,lspn));
+endfunction

libs/libcodec2/octave/plppe.m

+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot two sparse phase prediction error text files.
+% Generate data from print_pred_error, print_pred_error_sparse_wo_correction1 etc
+
+function plppe(ppe1_file, ppe2_file, f)
+  
+  ppe1 = load(ppe1_file);
+  ppe2 = load(ppe2_file);
+
+  std1 = std(nonzeros(ppe1(:,40:80)));
+  std2 = std(nonzeros(ppe2(:,40:80)));
+
+  printf("std dev for %s is %4.3f\n", ppe1_file, std1);
+  printf("std dev for %s is %4.3f\n", ppe2_file, std2);
+ 
+  figure(1);
+  clf;
+  subplot(211)
+  hist(nonzeros(ppe1(:,40:80)),20);
+  subplot(212)
+  hist(nonzeros(ppe2(:,40:80)),20);
+  
+  k = ' ';
+  do 
+    figure(2);
+    clf;
+    subplot(211)
+    L = length(nonzeros(ppe1(f,:)));
+    x = (1:L)*4000/L;
+    std1 = std(nonzeros(ppe1(f,:)));
+    legend = sprintf(";std dev %4.3f;", std1);
+    plot(x, nonzeros(ppe1(f,:)),legend);
+    axis([0 4000 -pi pi]);
+    subplot(212)
+    std2 = std(nonzeros(ppe2(f,:)));
+    legend = sprintf(";std dev %4.3f;", std2);
+    plot(x, nonzeros(ppe2(f,:)),legend);
+    axis([0 4000 -pi pi]);
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  p-png  q-quit ", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+    % optional print to PNG
+
+    if (k == 'p')
+       pngname = sprintf("%s_%d",samname,f);
+       png(pngname);
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/plsub.m

+% Copyright David Rowe 2010
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+% plots the difference of two files
+
+function plsub(samname1, samname2, start_sam, end_sam, pngname)
+  
+  fs1=fopen(samname1,"rb");
+  s1=fread(fs1,Inf,"short");
+  fs2=fopen(samname2,"rb");
+  s2=fread(fs2,Inf,"short");
+
+  st = 1;
+  en = length(s1);
+  if (nargin >= 3)
+    st = start_sam;
+  endif
+  if (nargin >= 4)
+    en = end_sam;
+  endif
+
+  figure(1);
+  clf;
+  l1 = strcat("r;",samname1,";");
+  plot(s1(st:en) - s2(st:en), l1);
+  %axis([1 en-st min(s1(st:en)) max(s1(st:en))]);
+ 
+  if (nargin == 5)
+    pngname = sprintf("%s.png",pngname);
+    print(pngname, '-dpng', "-S500,500")
+    pngname = sprintf("%s_large.png",pngname);
+    print(pngname, '-dpng', "-S800,600")
+  endif
+
+endfunction

libs/libcodec2/octave/plvoicing.m

+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot voicing information from sample and dump files.
+%
+% samfilename is the raw source file, e.g. "../raw/hts1a.raw"
+% samname is the dumpfile prefix, e.g. "../src/hts1a"
+%
+% There is a 160 sample (two frame delay) from the when a sample
+% enters the input buffer until it is at the centre of the analysis window
+
+function plvoicing(samfilename, samname, start_f, end_f, pngname)
+  
+  fs=fopen(samfilename,"rb");
+  s=fread(fs,Inf,"short");
+
+  snr_name = strcat(samname,"_snr.txt");
+  snr = load(snr_name);
+  model_name = strcat(samname,"_model.txt");
+  model = load(model_name);
+
+  Wo  = model((start_f+1):end_f,1);
+  F0  = Wo*4000/pi;
+  dF0 = F0(1:length(Wo)-1) - F0(2:length(Wo));
+
+  % work out LP and HP energy
+
+  for f=(start_f+1):end_f
+    L = model(f,2);
+    Am = model(f,3:(L+2));
+    L2 = floor(L/2);
+    elow = Am(1:L2) * Am(1:L2)';
+    ehigh = Am(L2:L) * Am(L2:L)';
+    erat(f-(start_f+1)+1) = 10*log10(elow/ehigh);
+  endfor
+
+  figure(1);
+  clf;
+  sp = s((start_f-2)*80:(end_f-2)*80);
+  plot(sp);
+  hold on;
+  vhigh = snr((start_f+1):end_f) > 7;
+  vlow = snr((start_f+1):end_f) > 4;
+
+  % test correction based on erat
+
+  vlowadj = vlow;
+
+  for f=1:length(erat)-1
+    if (vlow(f) == 0)
+      if (erat(f) > 10)
+        vlowadj(f) = 1;
+      endif
+    endif
+    if (vlow(f) == 1)
+      if (erat(f) < -10)
+        vlowadj(f) = 0;
+      endif
+      if (abs(dF0(f)) > 15)
+        vlowadj(f) = 0;
+      endif
+    endif
+  endfor
+
+  x = 1:(end_f-start_f);
+  plot(x*80,snr((start_f+1):end_f)*1000,';SNRdB x 1000;g+');
+  plot(x*80,-8000 + vhigh*2000,';7dB thresh;g');
+  plot(x*80,-11000 + vlowadj*2000,';vlow with corr;g');
+  plot(x*80,erat*1000,';elow/ehigh in dB;r');
+  plot(x*80,-14000 + vlow*2000,';4dB thresh;r');
+  hold off; 
+  grid
+  if (nargin == 5)
+      print(pngname, "-dpng", "-S500,500")
+  endif
+    
+  figure(2)
+  Wo  = model((start_f+1):end_f,1);
+  F0  = Wo*4000/pi;
+  dF0 = F0(1:length(Wo)-1) - F0(2:length(Wo));
+  %plot(dF0,'+--')
+  %hold on;
+  %plot([ 1 length(dF0) ], [10 10] ,'r')
+  %plot([ 1 length(dF0) ], [-10 -10] ,'r')
+  %axis([1 length(dF0) -50 50])
+  %hold off;
+  plot(F0,'+--')
+endfunction

libs/libcodec2/octave/sd.m

+% sd.m
+% David Rowe Aug 2012
+% Plots the spectal distorion between twofiles of LPCs.  Used for LSP 
+% quantisation tuning.
+
+function sd(raw_filename, dump_file_prefix, f)
+
+  ak1_filename = sprintf("%s_ak.txt", dump_file_prefix);
+  ak2_filename = sprintf("%s_ak_.txt", dump_file_prefix);
+  ak1 = load(ak1_filename);
+  ak2 = load(ak2_filename);
+
+  [ak1_r, ak1_c] = size(ak1);
+  [ak2_r, ak2_c] = size(ak1);
+
+  frames = max([ak1_r ak2_r]);
+  sd = zeros(1,frames);
+  Ndft = 512;
+  A1 = zeros(frames, Ndft);
+  A2 = zeros(frames, Ndft);
+
+  % initial helicopter view of all frames
+
+  for i = 1:frames
+    A1(i,:) = -20*log10(abs(fft(ak1(i,:),Ndft)));
+    A2(i,:) = -20*log10(abs(fft(ak2(i,:),Ndft)));
+    sd(i) = sum((A1(i,:) - A2(i,:)).^2)/Ndft;
+  end
+  printf("sd av %3.2f dB*dB\n", sum(sd)/frames);
+
+  figure(1);
+  clf;
+  subplot(211)
+  fs=fopen(raw_filename,"rb");
+  s = fread(fs,Inf,"short");
+  plot(s);
+  subplot(212)
+  plot(sd);
+
+  % now enter single step mode so we can analyse each frame
+
+  sn_name = strcat(dump_file_prefix,"_sn.txt");
+  Sn = load(sn_name);
+
+  lsp1_filename = sprintf("%s_lsp.txt", dump_file_prefix);
+  lsp2_filename = sprintf("%s_lsp_.txt", dump_file_prefix);
+  lsp1 = load(lsp1_filename);
+  lsp2 = load(lsp2_filename);
+
+  weights_filename = sprintf("%s_weights.txt", dump_file_prefix); 
+  if file_in_path(".",weights_filename)
+    weights = load(weights_filename);
+  end
+
+  k = ' ';
+  do 
+    figure(2);
+    clf;
+    subplot(211)
+    s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+    size(s);
+    plot(s);
+    axis([1 length(s) -20000 20000]);
+
+    subplot(212);
+    plot((1:Ndft/2)*4000/(Ndft/2), A1(f,1:(Ndft/2)),";A1;r");
+    axis([1 4000 -20 40]);
+    hold on;
+    plot((1:Ndft/2)*4000/(Ndft/2), A2(f,1:(Ndft/2)),";A2;");
+    if file_in_path(".",weights_filename)
+      plot(lsp1(f,:)*4000/pi, weights(f,:),";weights;g+");
+    end
+
+    for l=1:10
+        plot([lsp1(f,l)*4000/pi lsp1(f,l)*4000/pi], [0  -10], 'r');
+        plot([lsp2(f,l)*4000/pi lsp2(f,l)*4000/pi], [-10 -20], 'b');
+    endfor
+    plot(0,0,';lsp1;r');
+    plot(0,0,';lsp2;b');
+    sd_str = sprintf(";sd %3.2f dB*dB;", sd(f));
+    plot(0,0,sd_str);
+   
+    hold off;
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  q-quit", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction
+

libs/libcodec2/octave/spec.m

+% spec.m
+% Jean Marc Valin
+%
+% Spectrogram function for Octave 
+%
+%   Copyright (c) John-Marc Valin 2012
+%
+%   Redistribution and use in source and binary forms, with or without
+%   modification, are permitted provided that the following conditions
+%   are met:
+%   
+%   - Redistributions of source code must retain the above copyright
+%   notice, this list of conditions and the following disclaimer.
+%   
+%   - Redistributions in binary form must reproduce the above copyright
+%   notice, this list of conditions and the following disclaimer in the
+%   documentation and/or other materials provided with the distribution.
+%   
+%   - Neither the name of Jean Marc Valin nor the names of its
+%   contributors may be used to endorse or promote products derived from
+%   this software without specific prior written permission.
+%   
+%   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+%   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+%   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+%   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+%   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+%   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+%   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+%   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+%   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+%   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+%   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+function X = spec(x, Fs, framesize, start, stop)
+
+
+gr=[zeros(1024,1),[0:1023]'/1023,.68*[0:1023]'/1023];
+
+%gr=[.4*[0:1023]'/1023,[0:1023]'/1023,.68*[0:1023]'/1023];
+
+%t=[0:1023]'/1023;
+%t=(1+.25*t-cos(pi*t))/2.25;
+%gr = [zeros(1024,1),t,.68*t];
+
+
+%colormap(gr);
+
+if nargin < 2 || isempty(Fs)
+   Fs = 44100;
+end
+
+if nargin < 3 || isempty(framesize)
+   framesize = 2048;
+endif
+
+offset = framesize/4;
+
+X = 20*log10(abs(specgram(x, framesize, 48000, blackmanharris(framesize)', framesize-offset)));
+
+XM=max(max(X));
+X = max(XM-120,X);
+%size(X)
+F = -[framesize/2-1:-1:0]/framesize*Fs;
+%F = [0:127]/128*24000;
+T=[1:size(X,2)]/Fs*offset;
+%imagesc(X(end:-1:1,:));
+
+if nargin < 4 || isempty(start)
+   istart=1;
+else
+   istart = round(start*Fs/offset);
+end
+
+if nargin < 5 || isempty(stop)
+   istop = size(X,2);
+else
+   istop = round(stop*Fs/offset);
+endif
+
+istart = max(1,istart);
+istop = min(istop, size(X,2));
+
+imagesc(T(1+istart:istop), F, X(end:-1:1,1+istart:istop));
+
+X = X(:,1+istart:istop);

libs/libcodec2/octave/tget_spec.m

+% tget-spec.m
+%
+% Used in conjunction with src/fdmdv_demod to test the
+% fdmdv_get_rx_spectrum() function.
+%
+% codec2-dev/src$ ./fdmdv_demod fdmdv_mod.raw tmp.c2 dump.txt
+% octave:3> tget_spec("../src/dump.txt")
+%
+% Copyright David Rowe 2012
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+
+function tfft_log(dumpfilename)
+
+  load(dumpfilename);
+
+  [rows cols] = size(rx_spec_log_c);
+  Fs = 8000; low_freq = 0; high_freq = 2500;
+  res = (Fs/2)/cols;
+  st_bin = low_freq/res + 1;
+  en_bin = high_freq/res;
+  xaxis = (st_bin:en_bin)*res;
+
+  f_start = 2; f_end = 100;
+  beta = 0.1;
+
+  av = zeros(f_end, en_bin-st_bin+1);
+  for r=f_start:f_end
+      x = (1-beta)*av(r-1,:) + beta*rx_spec_log_c(r,st_bin:en_bin);
+      av(r,:) = x;
+  end
+
+  % spectrogram (waterfall)
+
+  figure(1)
+  clf;
+  imagesc(av,[-40 0]);
+
+  % animated spectrum display
+
+  figure(2)
+  clf;
+  for r=f_start:f_end
+      plot(xaxis, av(r,:))
+      axis([ low_freq high_freq -40 0])
+      sleep(0.1)
+  end
+endfunction

libs/libcodec2/octave/twotone.m

+% twotone.m
+% David Rowe Aug 2012
+% Used to experiment with combining phase of two tones
+
+function cbphase
+
+  Wo = 100.0*pi/4000;
+  L = floor(pi/Wo);
+  phi = zeros(1,L);
+
+  % two harmonics 
+
+  a = 20; b = 21;
+
+  % set up phases to whatever
+   
+  phi(a) = -pi;
+  phi(b) = -pi/2;
+
+  % synthesis the two-tone signal
+
+  N = 16000;
+  Nplot = 250;
+  s = zeros(1,N);
+
+  for m=a:b
+    s_m = cos(m*Wo*(0:(N-1)) + phi(m));
+    s = s + s_m;
+  endfor
+
+  % now our theory says that this signal should be the same perceptually
+
+  phi_(a) = (phi(a) - phi(b))/2;
+  phi_(b) = (phi(b) - phi(a))/2;
+
+  s_ = zeros(1,N);
+  for m=a:b
+    s_m = cos(m*Wo*(0:(N-1)) + phi_(m));
+    s_ = s_ + s_m;
+  endfor
+
+  % plot them and see if envelope has the same phase, but "carriers"
+  % have different phase
+
+  figure(1);
+  clf;
+  subplot(211);
+  plot(s(1:Nplot));
+  subplot(212);
+  plot(s_(1:Nplot),'r');
+endfunction
+

libs/libcodec2/octave/twotone1.m

+% twotone1.m
+% David Rowe Aug 17 2012
+%
+% Used to experiment with combining phase of two tones.  This version
+% sets up a complete synthetic speech signal then tries to combine the
+% phase of high frequency tones.  Lets see if we can do this and keep perceptual 
+% nature of signal the same.
+
+function twotone1
+ 
+  % hts1a frame 47
+
+  Wo = 0.093168;
+  L = 33;
+  %A = [69.626907 460.218536 839.677429 2577.498047 972.647888 712.755066 489.048553 364.830536 409.230652 371.767487 489.112854	893.127014 2447.596680 752.878113 475.720520 234.452271	248.161606 232.171051 202.669891 323.914490 678.749451 362.958038 211.652512 170.764435	148.631790 169.261673 272.254150 176.872375 67.344391 99.022301	60.812035 34.319073 14.864757];
+  A  = zeros(1,L)*100;
+  phi = [1.560274 1.508063 -1.565184 1.289117 -2.547365	1.412528 -1.303992 3.121130 1.087573 -1.158161 -2.928007 0.995093 -2.614023 0.246136 -2.267406 2.143802	-0.273431 -2.266897 1.685171 -0.668712 2.699722	-1.151891 2.406379 -0.046192 -2.718611 0.761067	-2.305014 0.133172 -1.428978 1.492630 -1.668385	1.539734 -1.336615];
+  %phi = zeros(1,L);
+  st = floor(L/2);
+  %st = 1;
+
+  A(st:st+5) = 1000;
+
+  % now set up phase of signal with phase of upper frequency harmonic
+  % pairs combined
+
+  phi_ = phi;
+  for m=floor(L/2):2:L
+    phi_(m)   = (phi(m) - phi(m+1))/2;
+    phi_(m+1) = (phi(m+1) - phi(m))/2;
+    %phi_(m+1) = 0;
+  end
+
+  % synthesise the signals
+
+  N = 16000;
+  Nplot = 250;
+
+  s = zeros(1,N);
+  for m=st:L
+    s_m = A(m)*cos(m*Wo*(0:(N-1)) + phi(m));
+    s = s + s_m;
+  endfor
+
+  s_ = zeros(1,N);
+  for m=st:L
+    s_m = A(m)*cos(m*Wo*(0:(N-1)) + phi_(m));
+    s_ = s_ + s_m;
+  endfor
+
+  % plot them, expect to see similar time domain waveforms
+
+  figure(1);
+  clf;
+  subplot(211);
+  plot(s(1:Nplot));
+  subplot(212);
+  plot(s_(1:Nplot),'r');
+
+  figure(2);
+  clf;
+  subplot(211);
+  plot(s(1:Nplot)-s_(1:Nplot));
+
+  % save to disk
+
+  gain = 1;
+  fs=fopen("twotone1_orig.raw","wb");
+  fwrite(fs,gain*s,"short");
+  fclose(fs);
+  fs=fopen("twotone1_comb.raw","wb");
+  fwrite(fs,gain*s_,"short");
+  fclose(fs);
+  
+endfunction
+