Add PTT to TX0

CHANGELOG.txt

@@ -1,3 +1,214 @@
+Quisk Version 4.2.54  August 2026
+=================================
+Quisk can now accept audio samples for Tx from TCI at rates 48000, 24000, 12000 and 8000. Please test.
+
+Quisk now includes program icons in the directory "quisk/icons". The Windows icon is quisk/icons/quisk.ico.
+On Windows, right-click the desktop icon and change the icon to <path-to-quisk>/quisk/icons/quisk.ico.
+The Linux icon is quisk/icons/quisk.svg. On Linux, change the quisk.desktop file to
+Icon=<path-to-quisk>/quisk/icons/quisk.svg.
+
+Quisk Version 4.2.53  June 2026
+===============================
+I added TCI support for Software Defined Connectors CW skimmer. Please test.
+
+This version of Quisk now works on Python 3.14 and higher. I changed to a new build system because
+the old one has been deprecated by Python. They no longer support running setup.py.
+The Linux version has a new makefile. The Windows version is installed by pip as usual.
+Please test.
+
+Quisk Version 4.2.52  May 2026
+==============================
+This is a fix for various TCI bugs. Please test with WSJT-X and let me know if all problems are solved.
+It is unlikely that other digital software will connect without problems, but please test and let me know.
+
+Quisk Version 4.2.51  May 2026
+==============================
+This version can use the TCI protocol to connect to WSJT-X. Please see Digital Modes in the documentation.
+
+Quisk Version 4.2.50  December 2025
+===================================
+This is a quick fix for a bug in 4.2.49 that results in the repeated message
+AttributeError: 'Hardware' object has no attribute 'is_HermesLite2'
+
+Quisk Version 4.2.49  December 2025
+===================================
+This version adds more features for the Anan 10E and other OpenHPSDR radios. Please test.
+
+I added the forward power, SWR, temperature and PA current display to the OpenHPSDR radio. But
+I don't have that radio so I need some help. I think the forward and reverse power connections
+are correct, but it is still necessary to create a table to convert analog input codes to power.
+Go to the Hardware screen and check the "Power meter calibration" setting. You will need to enter
+"New" and create a suitable table. To see the ADC values, choose table "Show ADC".
+
+There is no protocol standard to specify which analog input is used for temperature and PA current.
+And AFAIK the Anan 10E has no such sensors. I am hoping to use the board ID to find the right input.
+The ADC conversion is linear and the two points are specified on the Hardware screen.
+
+I currently assigned the analog inputs as follows:
+AIN1	Forward power
+AIN2	Reverse power
+AIN3	PA current
+AIN5	Temperature
+If you need to see the ADC codes 0-4095 for each input, go to hermes/quisk_hardware.py and
+change DEBUG_SENSORS to one.
+
+Quisk Version 4.2.48  November 2025
+===================================
+This is a quisk fix for a bug in the new OpenHPSDR radios. It fixes use_rx_udp that was reset to zero.
+
+Quisk Version 4.2.47  November 2025
+===================================
+To improve the performance of Quisk with the Anan 10-E and other OpenHPSDR radios, I added
+a new type of radio "OpenHPSDR". Use this type for the Anan 10-E and similar radios. It removes
+all the protocol extensions added by the Hermes-Lite2 project.  I removed the configuration
+option "Hermes Lite2 protocol enable". This is now automatic.
+
+I added buttons for antennas 0, 1, 2 and 3 for OpenHPSDR radios that support more antennas.
+I added controls for the OpenHPSDR preamplifier and attenuator.
+
+I do not have any Apache Labs radios, so I must depend on my users to report problems and solutions.
+
+Quisk Version 4.2.46  October 2025
+==================================
+I fixed a bug in Hamlib rig 2 which happens with the new Pure Signal code. The Hamlib "f" command
+now always returns the main receiver frequency, not one of the added receiver frequencies. This is
+consistent with the Flex ZZ commands.
+
+Quisk Version 4.2.45  October 2025
+==================================
+The new FreeDV including RADE mode is now a built-in mode of Quisk. Just press "FDV" to talk FreeDV.
+In brief, install FreeDV-GUI from the web site. Then configure the usual Quisk digital input and
+output that you use for digital modes. In FreeDV-GUI use the digital input and output. For the
+speakers and microphone use the same ones as Quisk. Don't forget to configure your call sign in
+FreeDV so it works with FreeDV Reporter. For detailed instructions see http://james.ahlstrom.name/rade.html.
+
+The Quisk digital output device is now always active; a DGT mode is not required.
+
+Quisk Version 4.2.44  September 2025
+====================================
+Quisk now has Pure Signal for the Hermes Lite2. This is baseband predistortion to reduce IMD products.
+To use it, press "PsCal" for calibration, and "PS" to enable Pure Signal. There are no adjustments needed.
+More details are in the documentation.
+
+Quisk Version 4.2.43  September 2025
+====================================
+I fixed a bug in the WDSP library NR2 noise reduction feature. I added the "Trained" method of NR2.
+Quisk uses the WDSP library for the NR2 noise reduction and the SNB noise blanker. The WDSP library must be available
+for these features to work. For windows, the library is included. For Linux you must build it by changing to the
+quisk/wdsp directory and entering "make". The library is used in-place and no installation is necessary.
+The new "trained" method of NR2 is included. Press the NR2 button twice to select "NR2t".
+
+I added documentation for the new FreeDV mode RADE. See http://james.ahlstrom.name/rade.html.
+The default IMD magnitude is now 100%.
+
+Quisk Version 4.2.42  April 2025
+================================
+I fixed a bug in the Kenwood IF CAT command. Thanks Nick!  I moved the Pulse server IP address to the Sound screen.
+I added a CAT method to imitate the Elecraft K4 command set over a TCP port. Quisk now has four CAT controls. To
+make this clearer I added more documentation to the Remote screen.
+
+Quisk Version 4.2.41  January 2025
+==================================
+I made some improvements to the top-level Config/Radios screen. There is now an option to
+make a new radio by copying settings from an existing radio. I worked around a bug in wxPython
+and gtk3 that caused drawing problems on the config screens. This problem is not present on Windows.
+
+Quisk Version 4.2.40  November 2024
+===================================
+Python 3.13 eliminated the telnetlib module, so I wrote a replacement. I fixed a bug in window sizing.
+If you use the Dx Cluster feature, please report any problems. Dx Cluster used to use the telnetlib module
+but is now new code.
+
+I updated the install to Python 3.13. Please report any problems with Python 3.13.
+
+Quisk Version 4.2.39  October 2024
+==================================
+I added an optional Hardware method PollGuiControl(self). If you define this in your Hardware file,
+the GUI thread will call it frequently. It can be used to set PTT from your hardware.
+
+I am trying to work around a wxPython quirk that causes mouse wheel tuning to fail. Please test.
+
+Quisk Version 4.2.38  September 2024
+====================================
+This is a bugfix release. I fixed a bug in the Hamlib serial port when using a physical TTY instead
+of a pseudo-tty. The Hamlib ZZFA command now changes the receive frequency instead of the transmit
+frequency. I fixed some window sizing issues.
+
+Quisk Version 4.2.37  August 2024
+=================================
+Quisk now has a new option "Fixed tune offset" on the Config/radio/Options screen. This is meant for
+Softrock radios that need a constant audio frequency to simplify amplitude and phase corrections.
+The value zero means no change to Quisk tuning. Otherwise enter the fixed frequency offset from center.
+If you are using the obsolete "mouse_tune_method = 1" please delete it and use this new feature.
+If you use softrock_tune_vfo.py as your hardware file, you should replace it with softrock/hardware_usb.py.
+
+Quisk now works on Python 3.12. And I fixed a bug in the small screen graph width.
+
+Quisk Version 4.2.36  August 2024
+=================================
+I added two new color schemes by Michael, DK1MI. The On/Off button color and text color can
+now be configured. (Michael, send me the final colors).  There are a few other small changes.
+
+Quisk Version 4.2.35  June 2024
+===============================
+There is a new item "File play level" on the Config/radio/Sound screen to reduce the volume when playing files.
+Use it to reduce the volume of the CQ message relative to radio sound.
+
+Dan, AI2M, provided a patch to compile in MacOS on Apple Silicon. Thanks Dan!
+
+I am working on better amplitude and phase adjustments when using SoftRock radios. The new corrections
+screen has a spreadsheet to display the VFO and tune offset. And there is a feature to automatically
+measure the corrections when using a signal generator as a source. If you use Quisk with SoftRock or similar
+radios, please let me know what you think.
+
+Quisk Version 4.2.34  May 2024
+==============================
+For the Hermes-Lite2 the inhibit Tx function (CN8) now releases the Spot button.
+
+Quisk displays the band plan below the X-axis as colored bars, and there are eighteen possible colors.
+If you hover the mouse over a bar, Quisk now displays the name of the mode. Recall that there is a dialog
+box so you can set the band plan to what you want.
+ 
+Quisk Version 4.2.33  May 2024
+==============================
+Added the get_level Hamlib command "l STRENGTH". It returns the S-meter reading. I finally found some Hamlib
+documentation, often not an easy task.  The return value of STRENGTH is in dB with zero dB equal to S9.
+There are 6 dB per S-unit. So a STRENGTH of 4 is 4dB over S9, and -6, -12, -18 is S8, S7, S6. I also added
+the corresponding FlexRadio command ZZSM.
+
+Quisk Version 4.2.32  April 2024
+================================
+I found and fixed a bug in the microphone filters that can result in a buffer overrun. You should update Quisk.
+
+Some Linux digital programs can not use the usual Quisk Pulse loopback device and must use an
+Alsa loopback. I added more documentation for Alsa loopbacks, added more information on the Sound
+screen and I changed the default Alsa sample size to 16 bit.
+
+Quisk now responds to the Hermes Lite2 Tx inhibit input by displaying a message and extinguishing the "Tx" light.
+There is a new item "Hermes Lite2 protocol" on the Hardware screen to control this behavior.
+
+Quisk Version 4.2.31  March 2024
+================================
+The "Tune" item was missing from the Midi menus. I fixed it but I didn't test it because I did
+not borrow Ben's Midi controller. Please test it yourself and let me know if it works. The Midi
+speed can be set from the menu too.
+
+Quisk Version 4.2.30  March 2024
+================================
+A new option "Tx-Rx silence msec" on the Timing configuration screen controls how long
+to suppress Rx audio when switching from Tx to Rx. The previous fixed time was 50 msec.
+
+The SSB squelch control now works on CW too. Please test.
+
+Quisk Version 4.2.29  January 2024
+==================================
+This version updates the FreeDV library files and program logic. I added a "Monitor"
+item to the FreeDV menu. It routes the sound to the speakers instead of to the demodulator.
+Turn on both the monitor and FDX to listen to your transmitted FreeDV signals. Please test.
+
+Modes 700x and 1600 are finished. The other modes will be added shortly. If you use any of them
+post here and I will move them to the top of the list.
+
 Quisk Version 4.2.28  December 2023
 ===================================
 This version includes a patch from Jon, AB8WU, to build Quisk on FreeBSD. Thanks Jon!

README.md

@@ -1,21 +0,0 @@
-# Quisk Software Defined Radio by N2ADR
-**December 26, 2023**
-
-This is Quisk, a Software Defined Radio (SDR).
-You supply radio hardware such as Hermes Lite2 or SoftRock to convert
-the antenna voltage to I/Q samples.
-Then send the samples to a computer running Quisk.
-The Quisk software will read the I/Q data, tune it, filter it,
-demodulate it, and send the audio to headphones or speakers.
-Quisk has a microphone input and a key input so it can operate as a
-complete transceiver.
-
-The web page for this project is  [https://groups.io/g/n2adr-sdr](https://groups.io/g/n2adr-sdr).
-
-The change log is [here](CHANGELOG.txt).
-
-The documentation is [here](docs.html).
-
-The help file is [here](help.html).
-
-This site is under construction. Check back soon.

__init__.py

@@ -1,2 +1,2 @@
-#Quisk version 4.2.29
-from .quisk import main
+#Quisk version 4.2.54
+from .quisk import main

ac2yd/control_common.py

@@ -395,7 +395,7 @@ class ControlCommon(BaseHardware):	# This is the Hardware class for the control
     for idName in ("Mute", "NR2", "AGC", "Sqlch", "NB 1", "Notch", "Test 1", "Spot", "FDX", "PTT", "VOX"):
       self.RemoteCtlSend("%s;%d\n" % (idName, app.idName2Button[idName].GetIndex()))
     # Menus
-    for menu in (app.NB_menu, app.split_menu, app.freedv_menu, app.smeter_menu):
+    for menu in (app.NB_menu, app.split_menu, app.smeter_menu):
       if menu:
         for nid in menu.id2data:
           menu_item = menu.FindItemById(nid)

ac2yd/remote.c

@@ -9,6 +9,7 @@
 #include <Python.h>
 #include <complex.h>
 #include <math.h>
+#include <string.h>
 #include <sys/time.h>
 #include <time.h>
 #include <errno.h>

afedrinet/afedrinet_io.c

@@ -3,6 +3,7 @@
 #ifdef MS_WINDOWS
 #include <Winsock2.h>
 #include <windows.h>
+#include <string.h>
 #else
 #include <sys/types.h>
 #include <sys/stat.h>
@@ -105,7 +106,7 @@ static PyObject * open_rx_udp(const char * ip, int port)
 	wVersionRequested = MAKEWORD(2, 2);
 	if (WSAStartup(wVersionRequested, &wsaData) != 0) {
 		sprintf(buf, "Failed to initialize Winsock (WSAStartup)");
-		return PyString_FromString(buf);
+		return PyUnicode_FromString(buf);
 	}
 #endif
 //	quisk_using_udp = 1;
@@ -145,7 +146,7 @@ static PyObject * open_rx_udp(const char * ip, int port)
 	else {
 		sprintf(buf, "Failed to open socket");
 	}
-	return PyString_FromString(buf);
+	return PyUnicode_FromString(buf);
 }
 
 static PyObject * close_rx_udp(PyObject * self, PyObject * args)
@@ -343,7 +344,7 @@ static PyObject * open_samples(PyObject * self, PyObject * args)
 	quisk_sample_source(NULL, NULL, &afedri_read_rx_udp);
 //////////////
 	return open_rx_udp(ip, port);		// AFEDRI specific
-//	return PyString_FromString(buf);		// return a string message
+//	return PyUnicode_FromString(buf);		// return a string message
 }
 
 // Miscellaneous functions needed by the SDR-IQ; called from the GUI thread as

afedrinet/makefile

@@ -1,7 +1,24 @@
+MODULE_NAME := afedrinet_io
+OBJDIR := ../build/local
 
-afedrinet2:
-	python2 setup.py build_ext --force --inplace
+CFLAGS			:= $(shell python3 ../makefil.py --cflags)
+LDFLAGS			:= $(shell python3 ../makefil.py --ldflags)
+EXTENSION_SUFFIX	:= $(shell python3 ../makefil.py --suffix)
+TARGET := $(MODULE_NAME)$(EXTENSION_SUFFIX)
 
-afedrinet3:
+# Default target: Link the shared library object
+$(TARGET): afedrinet_io.o $(OBJDIR)/import_quisk_api.o $(OBJDIR)/is_key_down.o
+	$(CC) -o $@ afedrinet_io.o $(LDFLAGS) $(OBJDIR)/import_quisk_api.o $(OBJDIR)/is_key_down.o
+
+# Compile the C source files into object files
+afedrinet_io.o: afedrinet_io.c ../quisk.h
+	$(CC) -o $@ -c $< $(CFLAGS) -I..
+
+.PHONY: clean
+clean:
+	-rm -f  *.o 
+	-rm -f  $(MODULE_NAME)*.so
+
+.PHONY: afedrinet-old
+afedrinet-old:
 	python3 setup.py build_ext --force --inplace
-

base64.c

@@ -0,0 +1,155 @@
+/*
+ * Base64 encoding/decoding (RFC1341)
+ * Copyright (c) 2005-2011, Jouni Malinen <j@w1.fi>
+ *
+ * This software may be distributed under the terms of the BSD license.
+ * See README for more details.
+ */
+
+#include <string.h>
+#include <stdlib.h>
+#include <stdint.h>
+#include "base64.h"
+
+static const unsigned char base64_table[65] =
+	"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
+
+/**
+ * base64_encode - Base64 encode
+ * @src: Data to be encoded
+ * @len: Length of the data to be encoded
+ * @out_len: Pointer to output length variable, or %NULL if not used
+ * Returns: Allocated buffer of out_len bytes of encoded data,
+ * or %NULL on failure
+ *
+ * Caller is responsible for freeing the returned buffer. Returned buffer is
+ * nul terminated to make it easier to use as a C string. The nul terminator is
+ * not included in out_len.
+ */
+unsigned char * base64_encode(const unsigned char *src, size_t len,
+			      size_t *out_len)
+{
+	unsigned char *out, *pos;
+	const unsigned char *end, *in;
+	size_t olen;
+	int line_len;
+
+	olen = len * 4 / 3 + 4; /* 3-byte blocks to 4-byte */
+	olen += olen / 72; /* line feeds */
+	olen++; /* nul termination */
+	if (olen < len)
+		return NULL; /* integer overflow */
+	out = malloc(olen);
+	if (out == NULL)
+		return NULL;
+
+	end = src + len;
+	in = src;
+	pos = out;
+	line_len = 0;
+	while (end - in >= 3) {
+		*pos++ = base64_table[in[0] >> 2];
+		*pos++ = base64_table[((in[0] & 0x03) << 4) | (in[1] >> 4)];
+		*pos++ = base64_table[((in[1] & 0x0f) << 2) | (in[2] >> 6)];
+		*pos++ = base64_table[in[2] & 0x3f];
+		in += 3;
+		line_len += 4;
+		if (line_len >= 72) {
+			*pos++ = '\n';
+			line_len = 0;
+		}
+	}
+
+	if (end - in) {
+		*pos++ = base64_table[in[0] >> 2];
+		if (end - in == 1) {
+			*pos++ = base64_table[(in[0] & 0x03) << 4];
+			*pos++ = '=';
+		} else {
+			*pos++ = base64_table[((in[0] & 0x03) << 4) |
+					      (in[1] >> 4)];
+			*pos++ = base64_table[(in[1] & 0x0f) << 2];
+		}
+		*pos++ = '=';
+		line_len += 4;
+	}
+
+	if (line_len)
+		*pos++ = '\n';
+
+	*pos = '\0';
+	if (out_len)
+		*out_len = pos - out;
+	return out;
+}
+
+
+/**
+ * base64_decode - Base64 decode
+ * @src: Data to be decoded
+ * @len: Length of the data to be decoded
+ * @out_len: Pointer to output length variable
+ * Returns: Allocated buffer of out_len bytes of decoded data,
+ * or %NULL on failure
+ *
+ * Caller is responsible for freeing the returned buffer.
+ */
+unsigned char * base64_decode(const unsigned char *src, size_t len,
+			      size_t *out_len)
+{
+	unsigned char dtable[256], *out, *pos, block[4], tmp;
+	size_t i, count, olen;
+	int pad = 0;
+
+	memset(dtable, 0x80, 256);
+	for (i = 0; i < sizeof(base64_table) - 1; i++)
+		dtable[base64_table[i]] = (unsigned char) i;
+	dtable['='] = 0;
+
+	count = 0;
+	for (i = 0; i < len; i++) {
+		if (dtable[src[i]] != 0x80)
+			count++;
+	}
+
+	if (count == 0 || count % 4)
+		return NULL;
+
+	olen = count / 4 * 3;
+	pos = out = malloc(olen);
+	if (out == NULL)
+		return NULL;
+
+	count = 0;
+	for (i = 0; i < len; i++) {
+		tmp = dtable[src[i]];
+		if (tmp == 0x80)
+			continue;
+
+		if (src[i] == '=')
+			pad++;
+		block[count] = tmp;
+		count++;
+		if (count == 4) {
+			*pos++ = (block[0] << 2) | (block[1] >> 4);
+			*pos++ = (block[1] << 4) | (block[2] >> 2);
+			*pos++ = (block[2] << 6) | block[3];
+			count = 0;
+			if (pad) {
+				if (pad == 1)
+					pos--;
+				else if (pad == 2)
+					pos -= 2;
+				else {
+					/* Invalid padding */
+					free(out);
+					return NULL;
+				}
+				break;
+			}
+		}
+	}
+
+	*out_len = pos - out;
+	return out;
+}

base64.h

@@ -0,0 +1,19 @@
+/*
+ * Base64 encoding/decoding (RFC1341)
+ * Copyright (c) 2005, Jouni Malinen <j@w1.fi>
+ *
+ * This software may be distributed under the terms of the BSD license.
+ * See README for more details.
+ */
+
+#ifndef BASE64_H
+#define BASE64_H
+
+#include <sys/types.h>
+
+unsigned char * base64_encode(const unsigned char *src, size_t len,
+			      size_t *out_len);
+unsigned char * base64_decode(const unsigned char *src, size_t len,
+			      size_t *out_len);
+
+#endif /* BASE64_H */

docs.html

@@ -73,6 +73,9 @@ h3     {font-size:1.6rem;}
 <li><a href="#Digital">Digital Modes</a></li>
 <li><a href="#Hamlib">Rig Control and Logging</a></li>
 <li><a href="#wsjtx">WSJT-X</a></li>
+<li><a href="#freedv">FreeDV</a></li>
+<li><a href="#wdsp">NR2 and WDSP</a></li>
+<li><a href="#pure">Pure Signal</a></li>
 <li><a href="#VNA">VNA</a></li>
 <li><a href="#Remote">Remote Operation</a></li>
 </ul>
@@ -81,7 +84,7 @@ h3     {font-size:1.6rem;}
 <div class="contents">
 <h1>
 <a id="Welcome"></a>
-Welcome to QUISK (December 2023)
+Welcome to QUISK (June 2026)
 </h1>
 
 <p>
@@ -131,7 +134,7 @@ button on the lower right.
 <h2>News</h2>
 <p>
 The newest versions of Linux and Raspberry Pi OS do not allow installing Quisk in the system Python package directory.
-I am working on a solution for this.
+Use the Linux Source Installation below instead.
 </p>
 <br>
 <p>
@@ -189,24 +192,23 @@ Many others contributed to Quisk, and are mentioned in comments in the source co
 <h2 id="Installation">Installation</h2>
 <p>
 Quisk is free open source software written in Python and C. It is hosted on the  PyPi repository
-<a href="https://pypi.org">https://pypi.org</a>.
-It can be installed using the standard Python setup tools. See specific directions below.
+<a href="https://pypi.org">https://pypi.org</a> and it is available on github.
 </p>
 <br>
 <p>
 If you have Python 3 installed on your computer, just continue to use that version.
-<b>Quisk requires Python 3.8 or 3.9 or 3.10 or 3.11.</b>
-If you want to install a more recent version, uninstall the old version first.
+On Windows Quisk requires Python 3.8 or higher.
+If you want to install a more recent version of Python, uninstall the old version first.
 Running multiple versions of Python is possible but can be confusing and is not necessary.
-A Quisk installation is needed for each version of Python you have.
+<b>A Quisk installation is needed for each version of Python you have. If you upgrade to a newer Python
+you must re-install Quisk. But your Quisk settings will be preserved.</b>
 </p>
 <br>
 <h3 id="g0.2.2">Windows Initial Installation</h3>
 <p>
-Windows does not include Python, so you must first install the most recent version of 64-bit Python 3.
+Windows does not include Python, so you must first install a recent version of 64-bit Python 3.
 If you already have 64-bit Python 3 installed, just keep using it.
-<b>Quisk requires Python 3.8 or 3.9 or 3.10 or 3.11.</b>
-There may be newer versions of Python, but until support is added, use the versions listed above.
+On Windows Quisk requires Python 3.8 or higher.
 Python is available from <a href="http://www.python.org/">http://www.python.org</a>.
 There is an option to <b>Add Python to PATH</b>. You MUST select
 this option so that you can start python by just typing "python" instead of the whole path
@@ -240,17 +242,26 @@ python -m pip install --upgrade pyserial
 <br><br>
 python -m pip install --upgrade quisk
 <br><br>
-You should then be able to start Quisk with the command "quisk".
-You can also start Quisk with "python -m quisk".
 To create a Quisk shortcut on your desktop, right-click an empty space and select "New" and "Shortcut".
-Use "quisk.exe" as the command and "Quisk" as the name.
-The "quisk.exe" program is in <b>your_install_directory</b>\Scripts.
+Enter this as the location of the item:
+</p>
+<br>
+<pre>
+pythonw -m quisk
+</pre>
+<br>
+<p>
+and then name the shortcut, perhaps "Quisk".
+If you check the properties you will see that Windows expanded the location and start directory to the full path.
 If you are curious, Quisk and its Python source files are installed in "<b>your_install_directory</b>\Lib\site-packages\quisk".
-If you change to that directory you can run Quisk with "python quisk.py".
+If you change to that directory you can run Quisk with "pythonw quisk.py".
+Quisk has its icon file in quisk\icons. Right-click your desktop shortcut and change the Icon to
+ "<b>your_install_directory</b>\Lib\site-packages\quisk\icons\quisk.ico".
 </p>
 <br>
 <p>
 To get started you must tell Quisk what kind of radio hardware you have. Press the Config button and select Radios.
+Add a radio from the list of hardware radios available. Or add the best match.
 Then set your sound devices for that radio; the device for the radio speakers, the microphone and so forth. All configuration
 is (mostly) from the Config button. Ignore old directions and don't bother with a config file.
 </p>
@@ -280,186 +291,115 @@ python -m pip uninstall quisk
 </p>
 <h3 id="g0.2.1">Linux Initial Installation</h3>
 <p>
-<br>
-The libsoundio-dev module is NO LONGER required!
-<br>
-<br>
 Linux runs on a wide variety of computers with different processors and versions of Linux.
 Therefore Quisk is compiled for each one. To do this, Python and a number of other packages are required.
-Most likely both Python2 and Python3 are already installed on your computer. Either the 32-bit or 64-bit
-version will work. On my Ubuntu machine, "python2" starts Python2 and "python3" starts Python3.
+Most likely both Python2 and Python3 are already installed on your computer.
+On my Ubuntu machine, "python2" starts Python2 and "python3" starts Python3.
 Just "python" starts Python2, but this will change as Python2 is phased out.
-Python2 is obsolete, so you should only install Quisk to Python3, and use "python3" to start it.
-Install these packages. Some of these can be installed from Python using "pip",
-but it is better to use your package manager because pip and the package manager do not coordinate.
+<b>Python2 is obsolete</b>, so you should only install Quisk to Python3, and use "python3" to start it.
+Install these packages by using your package manager, not Python pip.
 If newer versions of these packages become available, Linux will notify you.
-Use the most recent version of python3-wxgtk available.
+Use the most recent version of python3-wxgtk available for your Python version.
 <br><br>
 </p>
-<table>
-  <tr>
-    <th>Package Description</th>
-    <th>Python3 Package</th>
-  </tr>
-  <tr>
-    <td>Discrete fourier transform</td>
-    <td>sudo apt-get install libfftw3-dev</td>
-  </tr>
-  <tr>
-    <td>Alsa sound library</td>
-    <td>sudo apt-get install libasound2-dev</td>
-  </tr>
-  <tr>
-    <td>Portaudio sound library</td>
-    <td>sudo apt-get install portaudio19-dev</td>
-  </tr>
-  <tr>
-    <td>Pulseaudio sound library</td>
-    <td>sudo apt-get install libpulse-dev</td>
-  </tr>
-  <tr>
-    <td>Python development</td>
-    <td>sudo apt-get install python3-dev</td>
-  </tr>
-  <tr>
-    <td>Python library development</td>
-    <td>sudo apt-get install libpython3-dev</td>
-  </tr>
-  <tr>
-    <td>wxPython GUI library</td>
-    <td>sudo apt-get install python3-wxgtk4.0</td>
-  </tr>
-  <tr>
-    <td>Python USB</td>
-    <td>sudo apt-get install python3-usb</td>
-  </tr>
-  <tr>
-    <td>Python serial port</td>
-    <td>sudo apt-get install python3-serial</td>
-  </tr>
-  <tr>
-    <td>Python installation tools</td>
-    <td>sudo apt-get install python3-setuptools</td>
-  </tr>
-  <tr>
-    <td>Pip package installer</td>
-    <td>sudo apt-get install python3-pip</td>
-  </tr>
-</table>
 <p>
-<br><br>
+These are part of your operating system. You only install them once unless it changes.
+</p>
+<br>
+<pre>
+sudo apt-get install libfftw3-dev
+sudo apt-get install libasound2-dev
+sudo apt-get install portaudio19-dev
+sudo apt-get install libpulse-dev
+sudo apt-get install python3-dev
+sudo apt-get install libpython3-dev
+</pre>
+<br>
+<p>
+These are part of Python. You only install them once unless the Python started by "python3" changes.
+</p>
+<br>
+<pre>
+sudo apt-get install python3-wxgtk4.0
+sudo apt-get install python3-usb
+sudo apt-get install python3-serial
+sudo apt-get install python3-setuptools
+</pre>
+<br>
+<p>
 If you want to use the SoapySDR module make sure that SoapySDR is installed from its Pothosware site before you build Quisk.
 Otherwise Quisk will not have the files it needs. You can install SoapySDR later, but then you
-will need to reinstall Quisk. This is also true of any other external packages that have their own C source.
+will need to run "make" again. This is also true of any other external packages that have their own C source.
 <br>
 <br>
 </p>
 <h3>Linux Pip Installation</h3>
+<br>
 <p>
-You can use the Python program "pip" to install Quisk into the Python package directory. This is the easiest
-method, but it will not install the source files. If you want the source files see below.
-<br>
-<br>
-sudo -H python3 -m pip install --upgrade quisk
-<br>
-<br>
-This will install the Python files and all other files except for the C source. This is the easiest method
-if you do not want to work in C.
-To run Quisk, use this command from a terminal:
-<br>
-<br>
-python3 -m quisk
-<br>
-<br>
-You can create a panel launcher on your desktop with the command to run Quisk.
-<br>
-<br>
-If the installation of Quisk fails, you can force re-installation:
-<br>
-<br>
+<b> Installation of Quisk using Python tools like pip is deprecated</b>.
 </p>
-<pre>
-sudo -H python3 -m pip uninstall quisk
-sudo -H python3 -m pip install  --no-cache-dir  --force-reinstall  quisk
-</pre>
+<br>
 <p>
+Installation to the system Python is no longer allowed, and you may get the message "error: externally-managed-environment".
+If you used pip to install Quisk you can continue to upgrade with "sudo python3 -m pip install --upgrade quisk"
+until you get this error message. But it is best to uninstall the pip Quisk. If you installed with pipx, uninstall that too.
+</p>
 <br>
-You can also install an older version of Quisk. You may need to do that if the most recent version fails for some reason. Use:
+<pre>
+sudo python3 -m pip uninstall quisk
+pipx uninstall quisk
+</pre>
+<br>
+<p>
+Then use the Linux Source Installation.
+</p>
 <br>
 <br>
-sudo -H python3 -m pip install quisk==4.1.51
+<h3 id="LSI">Linux Source Installation</h3>
+<p>
+On Linux, Quisk is installed from github, and it is contained in a "quisk" directory. First you need to decide where to put the
+quisk directory. Anywhere will do, but it is best to use a directory where you have write permission so
+you don't need to use "sudo". The examples below assume you used your home directory indicated by "~". Just replace "~"
+by your desired location.
+</p>
 <br>
+<p>
+To install a fresh copy of Quisk, change to your desired directory, clone the github site, change to the quisk directory and run "make".
+Then run the python program quisk.py.
+</p>
 <br>
+<pre>
+cd ~
+git clone https://github.com/jimahlstrom/quisk
+cd quisk
+make
+python3 quisk.py
+</pre>
+<br>
+<p>
+If "make" fails, you probably have missing packages or missing "-dev" packages. Try to figure out what is missing from the error messages.
+You only clone github for the initial installation. To update Quisk, use pull followed by make:
+</p>
+<br>
+<pre>
+cd ~/quisk
+git pull
+make clean
+make
+</pre>
+<br>
+<p>
+You run Quisk by running the file quisk.py with python3. For example, "python3 /home/jim/quisk/quisk.py".
+You can create a desktop icon to do that, but the method depends on your version of Linux.
+The quisk icon is the file quisk/icons/quisk.svg.
+<b>Never run Quisk with root privelege by using sudo. Fix the permission problem instead.</b>
 To get started you must tell Quisk what kind of radio hardware you have. Press the Config button and select Radios.
+Add a radio from the list of hardware radios available. Or add the best match.
 Then set your sound devices for that radio; the device for the radio speakers, the microphone and so forth. All configuration
 is (mostly) from the Config button. Ignore old directions and don't bother with a config file.
-<br>
-<br>
-To edit the Quisk Python files you need to know where they are. To find out, import quisk and print it:
-<br>
-<br>
-</p>
-<pre>
-jim@IntelNUC:~$ python3
-Python 3.6.9 (default, Oct  8 2020, 12:12:24) 
-[GCC 8.4.0] on linux
-Type "help", "copyright", "credits" or "license" for more information.
---- import quisk
---- quisk
-module 'quisk' from '/usr/local/lib/python3.6/dist-packages/quisk/__init__.py'
---- 
-
-</pre>
-<br>
-<h3>Linux Pip Upgrade</h3>
-<p>
-To upgrade to a newer version of Quisk, use pip:
-<br>
-<br>
-sudo python3 -m pip install --upgrade quisk
-<br>
-<br>
-</p>
-<h3>Linux Uninstall Quisk</h3>
-<p>
-sudo python3 -m pip uninstall quisk
+To uninstall Quisk, just delete the quisk directory.
 </p>
 <br>
-<br>
-<h3>Linux Source Installation</h3>
-<p>
-The previous method using pip is simple, but will not install the C source files. If you want those proceed as follows.
-Go to <a href="https://pypi.org/project/quisk/">https://pypi.org/project/quisk</a> and under "Navigation"
-select "Download files". Download the quisk-x-x-xx.tar.gz file. This is the source file. The ".whl" files are for Windows.
-Change directory to where the file was downloaded, probably "Downloads", and uncompress and untar it.
-<br><br>
-cd  ~/Downloads
-<br>
-tar zxf quisk-x.x.xx.tar.gz
-<br><br>
-Change directories to the Quisk directory:
-<br><br>
-cd quisk-x.x.xx
-<br><br>
-Now compile Quisk with the "make" command:
-<br><br>
-make quisk3
-<br><br>
-If "make" fails, you probably have missing packages or missing "-dev" packages. Try to figure out what is missing from the error messages.
-Now you can run quisk with the command:
-<br><br>
-python3 quisk.py
-<br><br>
-At this point you have a choice of where to install Quisk. You could just rename the quisk-x.x.xx directory
-to a more convenient name, like "quisk" in your home directory, and run Quisk from there.
-This is convenient if you want to
-alter the Quisk code. Or you could install Quisk into the Python system.
-<br><br>
-python2 setup.py build
-<br>
-sudo python2 setup.py install
-<br><br>
-</p>
 <h3 id="g0.2.4">Quisk Files</h3>
 <p>
 <br>
@@ -602,7 +542,7 @@ When you first install Quisk, you will not have any settings for your radio.
 Press the Config button and go to the Radios screen.
 Then create a radio by specifying the general hardware type and give it a name of your choosing.
 For a Hermes-Lite, specify "Hermes" as the hardware type and call it "HL2" (or some other name).
-Press "Add" and a new tab for your radio will appear. Look through the various settings on the HL2 tab.
+Press "Apply" and a new tab for your radio will appear. Look through the various settings on the HL2 tab.
 The parameters for the radios are stored in the file quisk_settings.json.
 <br>
 <br>
@@ -900,8 +840,14 @@ SoftRock radios use an analog mixer to change the RF signal to stereo audio, and
 A high quality sound card is advisable. The analog mixer is not perfect, and it will be necessary to adjust the I and Q
 signals to equal amplitude and 90 degrees phase difference. Use the Config/Config screen buttons to
 bring up an adjustment screen. Adjustments must be made for each band, and separately for transmit and receive.
-The adjustment depends on both the VFO frequency and the tuning offset from the VFO. See my paper 
+The adjustment depends on both the VFO frequency and the tuning offset from the VFO.
+Quisk has an option to fix the tuning offset to a constant and to tune with the hardware.
+The tuning offset then has a constant correction and it is only necessary to correct for the VFO,
+a much less sensitive correction.
+See my paper 
 <a href="http://james.ahlstrom.name/phase_corr.html">http://james.ahlstrom.name/phase_corr.html</a> for more information.
+<br>
+<br>
 A good strategy is to pick a VFO near the band center, and record corrections at an Rx frequency of -15000, -1000, 1000 and 15000 Hertz.
 If the corrections are sensitive to VFO, record the corrections for these same Rx frequencies at VFOs equal to
 or slightly outside the upper and lower band edges.
@@ -910,15 +856,20 @@ You can add as many correction points as desired. The corrections are saved in t
 by hand if you are a Python expert. Otherwise you can just read the values.
 <br>
 <br>
+If you fix the tuning offset to a constant, you can create a correction for VFO at the band center, or
+the center and both edges.
+<br>
+<br>
 To create a Receive correction point for a given VFO and frequency, attach a signal generator to the SoftRock through an attenuator,
 and look at the image on the graph screen. If the signal is 3500 Hertz above the VFO, the image is 3500 Hertz below it.
-If you don't have a signal generator, find a strong station on the band and look at its image. Minimize the image by adjusting
-the amplitude and phase sliders on the adjustment screen, and then press "Save". 
+Choose "Measure" to measure the corrections and then "Add Cell".
+If you don't have a signal generator, find a strong station on the band and look at its image. Choose "Manual adjustment".
+Minimize the image by adjusting the amplitude and phase sliders on the adjustment screen, and then press "Add cell". 
 <br>
 <br>
 To create a Transmit correction point for a given VFO and frequency, attach the SoftRock RF output to a spectrum analyzer through an attenuator,
 and look at the image. If you don't have a spectrum analyzer use a second receiver tuned to the image. Minimize the image by adjusting
-the amplitude and phase sliders on the adjustment screen, and then press "Save".
+the amplitude and phase sliders on the adjustment screen, and then press "Add cell".
 <br>
 <br>
 </p>
@@ -1486,9 +1437,35 @@ is no longer needed
 <br>
 </p>
 <h2 id="Digital">Digital Modes </h2>
+<br>
+<h3>Digital Modes with the TCI Protocol</h3>
 <p>
+Using the sound devices in a PC to transfer digital sample streams has always caused problems.
+Several new methods are now becoming available. These are the <a href="https://github.com/ExpertSDR3/TCI">TCI protocol</a>
+used by <a href="https://eesdr.com/en/">Expert Electronics</a> for
+their SunSDR hardware
+and <a href="https://www.vita.com/page-1855484">Vita49</a> used by <a href="https://www.flexradio.com">FlexRadio</a>.
+And Hamlib is planning to add digital stream support.
+</p>
+<br>
+<p>
+Quisk can now connect to WSJT-X version 3.0.0 with the TCI protocol.
+On the Quisk Config/radio/Remote screen enter IP address 127.0.0.1 and port 40001.
+Use USB for digital modes. The DGT-* modes are for sound cards.
+In WSJT-X choose the radio "TCI Client RX1", and for audio choose "TCI audio".
+And enter the same IP and port. Everything then just works.
+If your WSJT-X does not have "TCI Client RX1" you need to update to a more recent version.
+This feature is new. To see the TCI conversation, use the "Debug level" on the Config/radio/Options screen.
+</p>
+<br>
+<h3>Digital Modes with Sound Cards</h3>
+<p>
+Digital modes need to receive the audio from the radio to decode it.
+They need to send audio to the radio to transmit.
+These audio streams have traditionally been sent using PC sound cards or
+an artificial sound card known as a Virtual Audio Cable.
 Quisk has a number of modes "DGT-" to receive and transmit digital
-signals. The modes "DGT-U" and "DGT-L" decode the signal as upper
+signals on sound devices. The modes "DGT-U" and "DGT-L" decode the signal as upper
 or lower sideband, and send the stereo audio to the digital sound
 device. The left and right channel are the same.
 The bandwidth is set with the filter buttons as
@@ -1544,28 +1521,47 @@ show the devices being used. Change the Fldigi playback to QuiskDigitalInput, an
 Monitor of QuiskDigitalOutput.  Then everything should work.  You do not need to use pavucontrol again because PulseAudio will
 remember the settings.
 <br>
-
 <br>
-If you don't have PulseAudio or don't want to use it, you can use the ALSA loopback device.
-The ALSA loopback device works the same way as the Windows VAC. First
-create the loopback device with the command "modprobe snd-aloop" (you
-will need to be root). You can create the loopback device
-when the system starts, but the way to do that depends on your version of
-Linux. I added snd-aloop to /etc/modules.  You could put the modprobe command in /etc/rc.local instead.
+Michael, DK1MI/N1BSD, has written a nice description of how to set up Fldigi. It is available
+at his site <a href="https://dk1mi.radio/quisk-fldigi/">https://dk1mi.radio/quisk-fldigi/</a>.
+<br>
+<br>
+If your digital program only uses Alsa devices or you don't want to use PulseAudio,
+you can use the ALSA loopback device.
+It works the same way as the Windows VAC. First create the loopback device with this command:
+<br>
+<br>
+<pre>
+sudo modprobe snd-aloop index=19
+</pre>
+<br>
+<p>
+The "index" is the card number, and it is best to use a number higher than your regular cards.
+The loopback device will be gone on the next restart.
+You can create the loopback device every time the system starts, but the way to do that depends on your version of Linux.
+You could add this line to /etc/modprobe.d/alsa-base.conf:
+</p>
+<br>
+<pre>
+options snd-aloop index=19
+</pre>
+<br>
+<p>
+Or you could add snd-aloop to /etc/modules or put the modprobe command in /etc/rc.local instead.
 Restart Linux. Now you can enter "cat /proc/asound/cards" to
-print out your sound cards, and you should see a "Loopback" card
-listed. The cards are also shown on the Quisk Sound
-screen. The Loopback card has one side that connects to Quisk and
+print out your sound cards, and you should see a "Loopback" card listed.
+The Loopback card has one side that connects to Quisk and
 another side that connects to your digital program. For the Quisk
-side connect both Digital Input and Digital Output to Loopback,0.
+side connect both Digital Input and Digital Output to the Loopback at hw:19,0.
 Note that the Loopback card is full duplex, and handles both
 input and output. There are actually eight loopbacks created at once,
 but we are only using subdevice 0. For the digital program side, set the input and
-output to "Loopback,1".
+output to the Loopback at hw:19,1.
 Your audio is now connected and you
 should be able to receive digital signals. Be sure to test your
 transmit signal off the air. You may need to reduce power to
 improve linearity.
+</p>
 <br>
 
 <br>
@@ -1641,6 +1637,46 @@ For Linux, use the pulseaudio devices:
   </tr>
 </table>
 <br><br>
+<h2 id="freedv">FreeDV Digital Voice</h2>
+<p>
+The <a href="https://freedv.org/">FreeDV project</a> has produced a new and exciting digital voice mode, the
+<a href="https://github.com/drowe67/radae/blob/main/doc/rade_intro_waveform.pdf">Radio Autoencoder RADE</a>. 
+The RADE mode is a real advance. It has a small peak to average power radio. And its digital constellation is not
+discrete dots but rather a continuous analog pattern. It works great and sounds great!
+You can make digital voice QSO's by using the FreeDV-GUI program. It runs in its own window and connects to your
+HF radio using the same virtual audio connections used by other digital programs such as WSJT-X.
+I wrote a <a href="http://james.ahlstrom.name/rade.html">guide to using FreeDV-GUI</a> to make it easier to install and use it.
+</p>
+<br><br>
+<h2 id="wdsp">NR2 and WDSP</h2>
+<p>
+Quisk uses the WDSP library for the NR2 noise reduction and the SNB noise blanker. The WDSP library must be available
+for these features to work. For windows, the library is included. For Linux you must build it by changing to the
+quisk/wdsp directory and entering "make". The library is used in-place and no installation is necessary.
+</p>
+<br>
+<p>
+The new "trained" method of NR2 is included. Press the NR2 button twice to select "NR2t".
+</p>
+<br><br>
+<h2 id="pure">Pure Signal</h2>
+<p>
+Pure Signal is a term used to describe baseband predistortion, a technique used to reduce IMD products
+caused by a non-linear power amp. Pure Signal is available for the Hermes Lite2.
+To use it first press the PsCal button to create a calibration. This will key a two second CW carrier.
+Then press PS to enable corrections. You can view the result by using the "FDX" button to see the Tx output.
+The feature is automatic and no settings are required. You can re-calibrate if you change bands, but this is not
+really necessary.
+</p>
+<br>
+<p>
+You need to supply a sample of the amplifier output so Pure Signal can correct it. First set the "LNA during Tx dB" to
+the minimum gain of -12 on the Hermes Hardware screen. This minimizes the preamp gain and the noise picked up from the HL2 board.
+Then provide a Tx sample much above the remaining board noise but below clipping. Use the FDX button to see the Tx output
+with and without the Tx sample connected. Twenty or thirty dB over board noise should be enough. To learn how to provide a Tx
+signal sample, see <a href="https://github.com/softerhardware/Hermes-Lite2/wiki/PureSignal">the HL2 Pure Signal wiki</a>.
+</p>
+<br><br>
 <h2 id="VNA">Vector Network Analyzer</h2>
 <p>
 If you have my transceiver hardware from 2010 QEX, or the newer HiQSDR

dxcluster.py

@@ -1,9 +1,9 @@
-from __future__ import absolute_import
 # This code was contributed by Christof, DJ4CM.  Many Thanks!!
+# Modified November 2024 by N2ADR to remove deprecated telnetlib module.
 
 import threading
+import socket
 import time
-import telnetlib
 import quisk_conf_defaults as conf
 
 class DxEntry:
@@ -81,61 +81,109 @@ class DxEntry:
   
 class DxCluster(threading.Thread):
   def __init__(self):
-    self.do_init = 1
+    self.error = 'Starting'
+    self.dxSpots = []
     threading.Thread.__init__(self)
     self.doQuit = threading.Event()
-    self.dxSpots = []
     self.doQuit.clear()
+    self.dxLock = threading.Lock()
+    self.addr = conf.dxClHost + ':' + str(conf.dxClPort)
+    self.msg_no_spots = "No DX Cluster data from " + self.addr
+    self.msg_one_spot = '1 DX spot received from ' + self.addr
+    self.msg_spots =    ' DX spots received from ' + self.addr
     
   def run(self):
-    self.telnetInit()
     self.telnetConnect()
+    if self.error:
+      self.sock.close()
+      return
     while not self.doQuit.isSet():
       try:
-          self.telnetRead()
+        by = self.sock.recv(1024)
+      except TimeoutError:
+        continue
       except:
-        self.tn.close()
-        time.sleep(20)
+        by = b''
+      if by:
+        with self.dxLock:
+          self.bytes += by
+      else:
+        self.sock.close()
+        self.error = "Restarting " + self.addr
+        time.sleep(2)
         if not self.doQuit.isSet():
           self.telnetConnect()
-    self.tn.close()
-      
-  def setListener (self, listener):  
-    self.listener = listener
+          if self.error:
+            self.sock.close()
+            return
+    self.sock.close()
         
-  def telnetInit(self):
-    self.tn = telnetlib.Telnet()
-      
   def telnetConnect(self):    
+    self.bytes = bytearray(0)
+    self.error = 'Starting'
+    self.sock = socket.socket()
+    self.sock.settimeout(20)
+    try:
+      self.sock.connect( (conf.dxClHost, conf.dxClPort) )
+    except:
+      self.error = "Failed to connect to " + self.addr
+      return
+    self.sock.settimeout(1)
     for i in range(10):
       try:
-        self.tn.open(conf.dxClHost, conf.dxClPort, 10)
-        self.tn.read_until(b"login:", 10)
-        self.tn.write(conf.user_call_sign.encode('utf-8', errors='ignore') + b"\n")		# user_call_sign may be Unicode
-        break
+        self.bytes += self.sock.recv(1024)
       except:
-        time.sleep(0.5)
+        pass
+      if b"login:" in self.bytes:
+        break
+    else:
+      self.error = 'No "login:" prompt from ' + self.addr
+      return
+    self.sock.sendall(conf.user_call_sign.encode('utf-8', errors='ignore') + b"\r\n")
     if conf.dxClPassword:
-      self.tn.read_until(b"Password: ")
-      self.tn.write(conf.dxClPassword.encode('utf-8', errors='ignore') + b"\n")
+      for i in range(10):
+        try:
+          self.bytes += self.sock.recv(1024)
+        except:
+          pass
+        if b"Password:" in self.bytes:
+          break
+      else:
+        self.error = 'No "Password:" prompt from ' + self.addr
+        return
+      self.sock.sendall(conf.dxClPassword.encode('utf-8', errors='ignore') + b"\r\n")
+    self.error = ''
+    self.bytes = bytearray(0)
 
-  def telnetRead(self):
-    message = self.tn.read_until(b'\n', 60).decode(encoding='utf-8', errors='replace')
-    if self.doQuit.isSet() == False:
-      dxEntry = DxEntry();
-      if dxEntry.parseMessage(message):
-        for i, listElement in enumerate(self.dxSpots):
-          if (listElement.equal(dxEntry)):
-            listElement.join (dxEntry)
-            return
-          if listElement.isExpired():
-            del (self.dxSpots[i])
-        self.dxSpots.append(dxEntry)
-        if self.listener:
-          self.listener()
-        
-  def getHost(self):
-    return self.tn.host + ':' + str(self.tn.port)
+  def Poll(self):
+    with self.dxLock:
+      index = self.bytes.find(b"\n")
+      if index >= 0:
+        message = self.bytes[0:index + 1]
+        self.bytes = self.bytes[index + 1:]
+      else:
+        return
+    message = message.decode(encoding='utf-8', errors='replace')
+    dxEntry = DxEntry()
+    if dxEntry.parseMessage(message):
+      for i, listElement in enumerate(self.dxSpots):
+        if (listElement.equal(dxEntry)):
+          listElement.join (dxEntry)
+          return True
+        if listElement.isExpired():
+          del (self.dxSpots[i])
+      self.dxSpots.append(dxEntry)
+      return True
         
   def stop(self):
     self.doQuit.set()
+
+  def dxStatus(self):
+    if self.error:
+      return self.error
+    nSpots = len(self.dxSpots)
+    if nSpots == 0:
+      return self.msg_no_spots
+    elif nSpots == 1:
+      return self.msg_one_spot
+    return str(nSpots) + self.msg_spots

extdemod.c

@@ -1,5 +1,6 @@
 #include <Python.h>
 #include <stdlib.h>
+#include <string.h>
 #include <math.h>
 #include <complex.h>
 #include "quisk.h"

filter.c

@@ -1,6 +1,7 @@
 #include <Python.h>
 #include <stdlib.h>
 #include <math.h>
+#include <string.h>
 #include <complex.h>	// Use native C99 complex type for fftw3
 #include "quisk.h"
 #include "filter.h"

filters.h

@@ -414,7 +414,7 @@ double quiskFilt111D4Coefs[196] = { 0.000022860383861326, 0.000070265173417164,
  0.000878636067621920, 0.000694724116305488, 0.000487289901409396, 0.000301317969124407, 0.000160689233204503,
  0.000070265173417164, 0.000022860383861326} ;
 
-// Sample 53333 Hz, pass 20000, stop 24000, ripple 0.1 dB, atten 100 dB.  For SDR-IQ.  Stop 0.45000.
+// Sample 53333 Hz, pass 20000, stop 24000, ripple 0.1 dB, atten 100 dB.  For SDR-IQ.  Pass 0.375. Stop 0.45000.
 double quiskFilt53D1Coefs[55] = { 0.000318858391362187, 0.001662116369387873, -0.000065237353639775, -0.000854885424783294,
  0.001693126335254083, -0.002034310867663472, 0.001354648789538254, 0.000509693525155950, -0.003079059747940221, 0.005242349925004462,
 -0.005626507739484119, 0.003235398612833476, 0.001883583294529978, -0.008273775245721798, 0.013282431836060808, -0.013924391581141530,
@@ -426,7 +426,7 @@ double quiskFilt53D1Coefs[55] = { 0.000318858391362187, 0.001662116369387873, -0
 -0.003079059747940221, 0.000509693525155950, 0.001354648789538254, -0.002034310867663472, 0.001693126335254083, -0.000854885424783294,
 -0.000065237353639775, 0.001662116369387873, 0.000318858391362187 } ;
 
-// Sample 53333 Hz, pass 10000, stop 12000, ripple 0.5 dB, atten 100 dB.  For SDR-IQ.  Stop 0.22500.
+// Sample 53333 Hz, pass 10000, stop 12000, ripple 0.5 dB, atten 100 dB.  For SDR-IQ.  Pass 0.1875. Stop 0.22500.
 double quiskFilt53D2Coefs[93] = { 0.000107966318729964, 0.000421475598772498, 0.000731343669484184, 0.000328611865414609,
 -0.001457272471508304, -0.004143803290279123, -0.005736802939367581, -0.004303861070721504, -0.000415676176666607,
  0.002561521351129964, 0.001705074918770091, -0.001794491466202784, -0.003398011312785532, -0.000595401366072839,
@@ -586,7 +586,7 @@ double quiskAudio24p6Coefs[36] = { 0.001199140008010727, 0.005953815908571521, 0
 -0.008973143363910490, 0.011839435384729619, 0.003122601697996235, -0.008665733839154772, -0.000319602525571569, 0.008621055763448699,
  0.005953815908571521, 0.001199140008010727} ;
 
-// Sample 48 kHz, pass 6, stop 8, ripple 0.5 dB, atten 80 dB.  Stop 0.16667.
+// Sample 48 kHz, pass 6, stop 8, ripple 0.5 dB, atten 80 dB.  Pass 0.125. Stop 0.16667.
 double quiskAudio48p6Coefs[71] = { 0.000324677779792651, 0.001038192405843851, 0.002082337464742640, 0.002938055116822665,
  0.002811674671240099, 0.001132278163758895, -0.001835394694865431, -0.004766409947196539, -0.005833007003128954, -0.003914015660218843,
  0.000296099005521212, 0.004239593489554206, 0.004918390640978499, 0.001100872619673516, -0.005241634309386784,
@@ -603,7 +603,7 @@ double quiskAudio48p6Coefs[71] = { 0.000324677779792651, 0.001038192405843851, 0
  0.001132278163758895, 0.002811674671240099, 0.002938055116822665, 0.002082337464742640, 0.001038192405843851,
  0.000324677779792651} ;
 
-// Sample 96 kHz, pass 6, stop 40, ripple 0.1 dB, atten 100 dB.  Stop 0.41667.
+// Sample 96 kHz, pass 6, stop 40, ripple 0.1 dB, atten 100 dB.  Pass 0.0625. Stop 0.41667.
 double quiskAudio96Coefs[11] = { -0.004756607210954994, -0.022893760473075838, -0.023609616122334920,
  0.078161168368381481, 0.277880235815005590, 0.388489826271597950, 0.277880235815005590,
  0.078161168368381481, -0.023609616122334920, -0.022893760473075838, -0.004756607210954994} ;
@@ -663,7 +663,7 @@ double quiskAudioFmHpCoefs[309] = { 0.004847574947800705, -0.001211970386842814,
 -0.000365311474947399, -0.000434052387522716, -0.000504808183463150, -0.000579117971409598, -0.000658731927598885, -0.000745515693376396,
 -0.000841696354699768, -0.000949749961336003, -0.001072277374239886, -0.001211970386842814, 0.004847574947800705} ;
 
-// Sample 24 kHz, pass 3, stop 4, ripple 0.1 dB, atten 100 dB.  Stop 0.166667.
+// Sample 24 kHz, pass 3, stop 4, ripple 0.1 dB, atten 100 dB.  Pass 0.125. Stop 0.166667.
 double quiskAudio24p3Coefs[100] = { 0.000040101217607380, 0.000144142945632828, 0.000313272623844654, 0.000472482653992188,
  0.000478145332060253, 0.000196098255702949, -0.000370577202968848, -0.000996638165520275, -0.001297816111521122,
 -0.000964536466628102, -0.000042557313824444, 0.000962997869327471, 0.001323534221298773, 0.000594056697947640,

freedv.c

@@ -1,707 +1 @@
-#include <Python.h>
-#include <stdlib.h>
-#include <math.h>
-#include <complex.h>	// Use native C99 complex type for fftw3
-#include <sys/types.h>
-
-#include "quisk.h"
-#include "freedv.h"
-#include <stdint.h>
-
-int DEBUG;
-
-#define MAX_RECEIVERS	2
-
-typedef struct {	// from comp.h
-  float real;
-  float imag;
-} COMP;
-
-struct freedv;		// from freedv_api.h
-typedef void (*freedv_callback_rx)(void *, char);
-typedef char (*freedv_callback_tx)(void *);
-/* Protocol bits are packed MSB-first */
-/* Called when a frame containing protocol data is decoded */
-typedef void (*freedv_callback_protorx)(void *, char *);
-/* Called when a frame containing protocol data is to be sent */
-typedef void (*freedv_callback_prototx)(void *, char *);
-/* Data packet callbacks */
-/* Called when a packet has been received */
-typedef void (*freedv_callback_datarx)(void *, unsigned char *packet, size_t size);
-/* Called when a new packet can be send */
-typedef void (*freedv_callback_datatx)(void *, unsigned char *packet, size_t *size);
-
-/* advanced freedv open options required by some modes */
-struct freedv_advanced {	// from freedv_api.h
-	int interleave_frames;
-};
-
-#ifdef MS_WINDOWS
-#include <windows.h>
-HMODULE hLib;
-#define GET_HANDLE1			hLib = LoadLibrary("libcodec2.dll")
-#define GET_HANDLE2			hLib = LoadLibrary(".\\freedvpkg\\libcodec2.dll")
-#define GET_HANDLE3			hLib = LoadLibrary(".\\freedvpkg\\libcodec2_32.dll")
-#define GET_HANDLE4			hLib = LoadLibrary(".\\freedvpkg\\libcodec2_64.dll")
-#define GET_ADDR(name)		(void *)GetProcAddress(hLib, name)
-#define CLOSE_LIB			FreeLibrary(hLib)
-#else
-#include <dlfcn.h>
-void * hLib;
-#define GET_HANDLE1			hLib = dlopen("libcodec2.so", RTLD_LAZY)
-#define GET_HANDLE2			hLib = dlopen("./freedvpkg/libcodec2.so", RTLD_LAZY)
-#define GET_HANDLE3			hLib = dlopen("./freedvpkg/libcodec2_32.so", RTLD_LAZY)
-#define GET_HANDLE4			hLib = dlopen("./freedvpkg/libcodec2_64.so", RTLD_LAZY)
-#define GET_ADDR(name)		dlsym(hLib,  name)
-#define CLOSE_LIB			dlclose(hLib)
-#endif
-
-static int requested_mode = -1;			// requested mode
-int freedv_current_mode = -1;			// the current running mode
-static int quisk_freedv_squelch;
-static int interleave_frames = 1;
-static int freedv_version = -1;
-static int quisk_set_tx_bpf = 1;
-static int handle_index = -1;
-// need access to these here and in quisk.c
-int n_speech_sample_rate = 8000;
-int n_modem_sample_rate = 8000;
-
-#define SPEECH_BUF_SIZE		12000		// speech buffer size increased from 3000
-static struct _rx_channel{
-	struct freedv * hFreedv;
-	COMP * demod_in;
-	int rxdata_index;
-	short speech_out[SPEECH_BUF_SIZE];		// output buffer
-	int speech_available;					// number of samples in output buffer
-	int playing;							// are we currently returning speech samples?
-} rx_channel[MAX_RECEIVERS] = { 0 };		// safe initialisation to 0
-
-// freedv_version is the library version number, or
-//   -1		no library was found
-//   -2		a library was found, but freedv_get_version is missing
-
-// FreeDV API functions:
-// open, close
-struct freedv * (*freedv_open)(int mode);
-struct freedv * (*freedv_open_advanced)(int mode, struct freedv_advanced *adv);
-void (*freedv_close)(struct freedv *freedv);
-// Transmit
-void (*freedv_tx)(struct freedv *freedv, short *, short *);
-void (*freedv_comptx)(struct freedv *freedv, COMP *, short *);
-// Receive
-int (*freedv_nin)(struct freedv *freedv);
-int (*freedv_rx)(struct freedv *freedv, short *, short demod_in[]);
-int (*freedv_floatrx)(struct freedv *freedv, short *, float demod_in[]);
-int (*freedv_comprx)(struct freedv *freedv, short *, COMP demod_in[]);
-// Set parameters
-void (*freedv_set_callback_txt)		(struct freedv *freedv, freedv_callback_rx rx, freedv_callback_tx tx, void *callback_state);
-void (*freedv_set_test_frames)			(struct freedv *freedv, int test_frames);
-void (*freedv_set_smooth_symbols)		(struct freedv *freedv, int smooth_symbols);
-void (*freedv_set_squelch_en)			(struct freedv *freedv, int squelch_en);
-void (*freedv_set_snr_squelch_thresh)	(struct freedv *freedv, float snr_squelch_thresh);
-void (*freedv_set_tx_bpf)		(struct freedv *freedv, int val);
-// Get parameters
-int (*freedv_get_version)(void);
-void (*freedv_get_modem_stats)(struct freedv *freedv, int *sync, float *snr_est);
-int (*freedv_get_test_frames)			(struct freedv *freedv);
-int (*freedv_get_n_speech_samples)		(struct freedv *freedv);
-int (*freedv_get_n_max_modem_samples)	(struct freedv *freedv);
-int (*freedv_get_n_nom_modem_samples)	(struct freedv *freedv);
-int (*freedv_get_total_bits)			(struct freedv *freedv);
-int (*freedv_get_total_bit_errors)		(struct freedv *freedv);
-int (*freedv_get_modem_sample_rate)		(struct freedv *freedv);
-int (*freedv_get_speech_sample_rate)	(struct freedv *freedv);
-//
-int (*freedv_get_sync)					(struct freedv *freedv);
-void (*freedv_set_callback_protocol)	(struct freedv *freedv, freedv_callback_protorx rx, freedv_callback_prototx tx, void *callback_state);
-void (*freedv_set_callback_data)	(struct freedv *freedv, freedv_callback_datarx datarx, freedv_callback_datatx datatx, void *callback_state);
-
-//
-// checkAvxSupport
-//
-// This function was copied from freedv, file fdmdv2_main.cpp and tweaked for use here
-//
-// Tests the underlying platform for AVX support.  2020 needs AVX support to run
-// in real-time, and old processors do not offer AVX support
-//
-
-#if defined(__x86_64__) || defined(_M_X64) || defined(__i386) || defined(_M_IX86)
-#include <cpuid.h>
-static int checkAvxSupport(void)
-{
-
-    int isAvxPresent = 0;
-    uint32_t eax, ebx, ecx, edx;
-    eax = ebx = ecx = edx = 0;
-    __cpuid(1, eax, ebx, ecx, edx);
-
-    if (ecx & (1<<27) && ecx & (1<<28)) {
-        // CPU supports XSAVE and AVX
-        uint32_t xcr0, xcr0_high;
-        asm("xgetbv" : "=a" (xcr0), "=d" (xcr0_high) : "c" (0));
-        isAvxPresent = (xcr0 & 6) == 6;    // AVX state saving enabled?
-    }
-	return isAvxPresent;
-}
-#else
-static int checkAvxSupport(void)
-{
-    return 0;
-}
-#endif
-
-/* Called when a new packet can be sent */
-void my_datatx(void *callback_state, unsigned char *packet, size_t *size) {
-	*size = 0;
-}
-
-static void GetAddrs(void)
-{
-	handle_index = 1;
-	GET_HANDLE1;
-	if ( ! hLib) {
-		handle_index = 2;
-		GET_HANDLE2;
-		if ( ! hLib) {
-			handle_index = 3;
-			GET_HANDLE3;
-			if ( ! hLib) {
-				handle_index = 4;
-				GET_HANDLE4;
-			}
-		}
-	}
-	if (hLib) {
-		freedv_get_version = GET_ADDR("freedv_get_version");
-		if (freedv_get_version != NULL) {
-			freedv_version = freedv_get_version();
-			if (DEBUG)
-				printf("FreeDV codec2 library %d found, version %d\n", handle_index, freedv_version);
-		}
-		else {
-			freedv_version = -2;
-			if (DEBUG)
-				printf("FreeDV codec2 library %d found, but no freedv_get_version() !!!\n", handle_index);
-			CLOSE_LIB;
-			return;
-		}
-	}
-	else {
-		handle_index = -1;
-		if (DEBUG)
-			printf("Could not find the FreeDV codec2 shared library\n");
-		return;
-	}
-
-// open, close
-	freedv_open = GET_ADDR("freedv_open");
-	freedv_open_advanced = GET_ADDR("freedv_open_advanced");
-	freedv_close = GET_ADDR("freedv_close");
-// Transmit
-	freedv_tx = GET_ADDR("freedv_tx");
-	freedv_comptx = GET_ADDR("freedv_comptx");
-// Receive
-	freedv_nin = GET_ADDR("freedv_nin");
-	freedv_rx = GET_ADDR("freedv_rx");
-	freedv_floatrx = GET_ADDR("freedv_floatrx");
-	freedv_comprx = GET_ADDR("freedv_comprx");
-// Set parameters
-	freedv_set_callback_txt = GET_ADDR("freedv_set_callback_txt");
-	freedv_set_callback_protocol = GET_ADDR("freedv_set_callback_protocol");
-	freedv_set_callback_data = GET_ADDR("freedv_set_callback_data");
-	freedv_set_test_frames = GET_ADDR("freedv_set_test_frames");
-	freedv_set_smooth_symbols = GET_ADDR("freedv_set_smooth_symbols");
-	freedv_set_squelch_en = GET_ADDR("freedv_set_squelch_en");
-	freedv_set_snr_squelch_thresh = GET_ADDR("freedv_set_snr_squelch_thresh");
-	freedv_set_tx_bpf = GET_ADDR("freedv_set_tx_bpf");
-// Get parameters
-	freedv_get_modem_stats = GET_ADDR("freedv_get_modem_stats");
-	freedv_get_test_frames = GET_ADDR("freedv_get_test_frames");
-	freedv_get_n_speech_samples = GET_ADDR("freedv_get_n_speech_samples");
-	freedv_get_n_max_modem_samples = GET_ADDR("freedv_get_n_max_modem_samples");
-	freedv_get_n_nom_modem_samples = GET_ADDR("freedv_get_n_nom_modem_samples");
-	freedv_get_total_bits = GET_ADDR("freedv_get_total_bits");
-	freedv_get_total_bit_errors = GET_ADDR("freedv_get_total_bit_errors");
-	freedv_get_sync = GET_ADDR("freedv_get_sync");		// requires version 11
-	if( freedv_version >= 11)
-		freedv_get_modem_sample_rate = GET_ADDR("freedv_get_modem_sample_rate");		// requires version 11 or later
-	if (freedv_version >= 12) 
-		freedv_get_speech_sample_rate = GET_ADDR("freedv_get_speech_sample_rate");		// requires version 12
-	return;
-}
-
-static int quisk_freedv_rx(complex double * cSamples, double * dsamples, int count, int bank)	// Called from the sound thread.
-{	// Input digital modulation is cSamples; decoded voice is dsamples.  Each "bank" is a stream of audio.
-	// caller will have decimated input audio down to n_modem_sample_rate and reduced block size to match reduction
-	// output data will be returned at n_speech_sample_rate with corresponding block sizes
-	int i, nout, need, have, sync;
-	int n_speech_samples;
-	complex double cx;
-	double scale = (double)CLIP32 / CLIP16;	// convert 32 bits to 16 bits
-	struct freedv * hF;
-	struct _rx_channel * pCh;
-	int nCountOut = 1;	// In case we have a higher (or lower) value for o/p count
-
-	if (cSamples == NULL) {		// shutdown
-		for (i = 0; i < MAX_RECEIVERS; i++) {
-			if (rx_channel[i].demod_in) {
-				free(rx_channel[i].demod_in);
-				rx_channel[i].demod_in = NULL;
-			}
-		}
-		return 0;
-	}
-
-	if (bank < 0 || bank >= MAX_RECEIVERS)
-		return 0;
-	hF = rx_channel[bank].hFreedv;
-	if ( ! hF)
-		return 0;
-	pCh = rx_channel + bank;
-	n_speech_samples = freedv_get_n_speech_samples(hF);
-	// if (say) input modem rate is the 2020 mode rate of 8000 sa/sec
-	// and output rate is the corresponding 16000 sa/sec then for an input
-	// block size of N, outbut 2N samples block size.
-	// Apply a multiplier/divisor to 'count' for the output samples count
-	if(n_speech_sample_rate >= n_modem_sample_rate ) {
-		int nRatio = n_speech_sample_rate / n_modem_sample_rate;
-		if( nRatio >= 1 && nRatio <= 6)
-			nCountOut = nRatio * count;
-	}
-	else {
-		int nRatio = n_modem_sample_rate / n_speech_sample_rate;
-		if( nRatio >= 1 && nRatio <= 6)
-			nCountOut = count / nRatio;
-	}
-	nout = 0;
-	need = freedv_nin(hF);
-	for (i = 0; i < count; i++) {
-		cx = cRxFilterOut(cSamples[i], bank, 0);
-		if (rxMode == FDV_L)		// lower sideband
-			cx = conj(cx);
-#if 1 // Try it the other way and remove scaling factor
-		pCh->demod_in[pCh->rxdata_index].real = creal(cx);
-		pCh->demod_in[pCh->rxdata_index].imag = cimag(cx);
-#else
-		pCh->demod_in[pCh->rxdata_index].real = (creal(cx) - cimag(cx));
-		pCh->demod_in[pCh->rxdata_index].imag = 0;
-#endif
-		pCh->rxdata_index++;
-		if (pCh->rxdata_index >= need) {
-			if (pCh->speech_available + n_speech_samples < SPEECH_BUF_SIZE) {		// check for buffer space
-				have = freedv_comprx(hF, pCh->speech_out + pCh->speech_available, pCh->demod_in);
-				if (freedv_version > 10)
-					sync = freedv_get_sync(hF);
-				else
-					freedv_get_modem_stats(hF, &sync, NULL);
-				if (freedv_current_mode == 0) {		// mode 1600
-					if (sync)		// throw away speech if not in sync
-						pCh->speech_available += have;
-				}
-				else if (pCh->speech_available < SPEECH_BUF_SIZE * 2 / 3) {
-					pCh->speech_available += have;	// keep speech if there is space
-				}
-				else {
-					if (DEBUG) printf("Close to maximum in speech output buffer\n");
-				}
-			}
-			else {		// no space in buffer
-				if (DEBUG) printf("Overflow in speech output buffer\n");
-			}
-			pCh->rxdata_index = 0;
-			need = freedv_nin(hF);
-		}
-	}
-	if ( ! pCh->playing) {
-		if (pCh->speech_available >= 2 * n_speech_samples) {
-			pCh->playing = 1;
-		}
-		else {		// return zero samples
-			for (i = 0; i < nCountOut; i++)
-				dsamples[i] = 0;
-			//if (DEBUG) printf("Rx buffer playing %d available %d\n", pCh->playing, pCh->speech_available);
-			return nCountOut;
-		}
-	}
-	for (nout = 0; nout < pCh->speech_available && nout < nCountOut; nout++)
-		dsamples[nout] = pCh->speech_out[nout] * scale * 0.7;
-	if (nout) {
-		pCh->speech_available -= nout;
-		memmove(pCh->speech_out, pCh->speech_out + nout, (pCh->speech_available) * sizeof(short));
-	}
-	if ( ! pCh->speech_available) {
-		pCh->playing = 0;
-		while (nout < nCountOut)
-			dsamples[nout++] = 0;
-	}
-	//if (DEBUG) printf("Rx buffer playing %d available %d\n", pCh->playing, pCh->speech_available);
-	return nout;
-}
-
-static int quisk_freedv_tx(complex double * cSamples, double * dsamples, int count)	// Called from the sound thread.
-{	// Input voice samples are dsamples; output digital modulation is cSamples.
-	int i, nout;
-	int n_speech_samples;
-	int n_nom_modem_samples;
-	static COMP * mod_out = NULL;
-	static short * speech_in = NULL;
-	static int speech_index=0, mod_index=0;
-	// g8kbb - add for 800XA
-	static short *real_mod_out = NULL;
-
-	if (dsamples == NULL) {		// shutdown
-		if (mod_out)
-			free(mod_out);
-		mod_out = NULL;
-		if (real_mod_out)
-			free(real_mod_out);
-		real_mod_out = NULL;
-		if (speech_in)
-			free(speech_in);
-		speech_in = NULL;
-		return 0;
-	}
-	if ( ! rx_channel[0].hFreedv)
-		return 0;
-	n_speech_samples = freedv_get_n_speech_samples(rx_channel[0].hFreedv);
-	n_nom_modem_samples = freedv_get_n_nom_modem_samples(rx_channel[0].hFreedv);
-	// find if need to output at faster rate
-	int nRatio = n_modem_sample_rate / n_speech_sample_rate;
-	if( nRatio < 1 || nRatio > 6 )
-		nRatio = 1;
-	if (mod_out == NULL) {		// initialize
-		mod_out = (COMP *)malloc(sizeof(COMP) * n_nom_modem_samples);
-		memset(mod_out, 0, sizeof(COMP) * n_nom_modem_samples);
-		speech_in = (short*)malloc(sizeof(short) * n_speech_samples);
-		speech_index=0;
-		mod_index=0;
-		real_mod_out = (short *)malloc(sizeof(short) * n_nom_modem_samples);
-		memset(real_mod_out, 0, sizeof(short) * n_nom_modem_samples);
-		//if( DEBUG ) printf("Initialise tx buffer size = %d, i/p sample size = %d\n", n_nom_modem_samples, n_speech_samples );
-	}
-	nout = 0;
-	for (i = 0; i < count; i++) {
-		speech_in[speech_index++] = (short)dsamples[i];
-		if (speech_index >= n_speech_samples) {
-			// Calculate a new block, but first write out the rest of the old block
-			// 800xa uses real not complex
-			if( freedv_current_mode == FREEDV_MODE_800XA ) {
-				for ( ; mod_index < n_nom_modem_samples; mod_index++)
-					cSamples[nout++] = real_mod_out[mod_index];
-				freedv_tx(rx_channel[0].hFreedv, real_mod_out, speech_in);
-			}
-			else {
-				for ( ; mod_index < n_nom_modem_samples; mod_index++)
-					cSamples[nout++] = mod_out[mod_index].real + I * mod_out[mod_index].imag;
-				freedv_comptx(rx_channel[0].hFreedv, mod_out, speech_in);
-			}
-			//if( DEBUG ) printf("Block processed, speech index=%d, n_speech_samples=%d, count=%d\n", speech_index, n_speech_samples, count );
-			mod_index = 0;
-			speech_index = 0;
-		}
-		else {		// write out samples
-			for( int j=0; j<nRatio; j++) {
-				if (mod_index < n_nom_modem_samples ) {
-					if( freedv_current_mode == FREEDV_MODE_800XA ) {
-						cSamples[nout++] = real_mod_out[mod_index];
-					}
-					else {
-						cSamples[nout++] = mod_out[mod_index].real + I * mod_out[mod_index].imag;
-					}
-					mod_index++;
-				}
-			}
-		}
-	}
-	if (rxMode == FDV_L)
-		for (i = 0; i < nout; i++)
-			cSamples[i] = conj(cSamples[i]);
-	return nout;
-}
-
-#define TX_MSG_SIZE		80
-static char quisk_tx_msg[TX_MSG_SIZE];
-
-static char get_next_tx_char(void * callback_state)
-{
-	char c;
-	static int index = 0;
-
-	c = quisk_tx_msg[index++];
-	if (index >= TX_MSG_SIZE)
-		index = 0;
-	if ( ! c) {
-		index = 0;
-		c = quisk_tx_msg[index++];
-	}
-	return c;
-}
-
-#define RX_MSG_SIZE		80
-static char quisk_rx_msg[RX_MSG_SIZE + 1];
-
-static void put_next_rx_char(void * callback_state, char ch)
-{
-	if (ch == '\n' || ch == '\r')
-		ch = ' ';
-	if (ch < 32 || ch > 126)	// printable characters
-		return;
-	if (strlen(quisk_rx_msg) < RX_MSG_SIZE)
-		strncat(quisk_rx_msg, &ch, 1);
-}
-
-PyObject * quisk_freedv_get_rx_char(PyObject * self, PyObject * args)	// Called from the GUI thread.
-{
-	PyObject * txt;
-
-	if (!PyArg_ParseTuple (args, ""))
-		return NULL;
-	txt = PyString_FromString(quisk_rx_msg);
-	quisk_rx_msg[0] = 0;
-	return txt;
-}
-
-static void CloseFreedv(void)	// Called from the GUI thread or sound thread
-{
-	int i;
-
-	for (i = 0; i < MAX_RECEIVERS; i++) {
-		if (rx_channel[i].hFreedv) {
-			freedv_close(rx_channel[i].hFreedv);
-			rx_channel[i].hFreedv = NULL;
-		}
-#if 0 // g8kbb - unnecessary - quisk_freedv_rx() will do it
-		if (rx_channel[i].demod_in) {
-			free(rx_channel[i].demod_in);
-			rx_channel[i].demod_in = NULL;
-		}
-#endif
-	}
-	quisk_freedv_rx(NULL, NULL, 0, 0);
-	quisk_freedv_tx(NULL, NULL, 0);
-	freedv_current_mode = -1;
-}
-
-static int OpenFreedv(void)	// Called from the GUI thread or sound thread
-{
-	int i, n_max_modem_samples;
-	struct freedv * hF;
-
-	if ( ! hLib)
-		GetAddrs();		// Get the entry points for funtions in the codec2 library
-	if (DEBUG) {	
-		switch (handle_index) {
-		case 1:
-			printf("freedv_open: system codec2 library found, version %d\n", freedv_version);
-			break;
-		case 2:
-			printf("freedv_open: codec2 library freedvpkg/libcodec2.dll|so found, version %d\n", freedv_version);
-			break;
-		case 3:
-		case 4:
-			printf("freedv_open: codec2 library freedvpkg/libcodec2_32|64.dll|so found, version %d\n", freedv_version);
-			break;
-		default:
-			printf("freedv_open: Could not find the FreeDV codec2 shared library\n");
-			break;
-		}
-	}
-	if (freedv_version < 10) {
-		CloseFreedv();
-		requested_mode = -1;
-		if (DEBUG)
-			printf("freedv_open: Failure because version is less than 10\n");
-		return 0;	// failure
-	}
-	if (requested_mode == FREEDV_MODE_2020) {
-		if (checkAvxSupport() == 0) {
-			CloseFreedv();
-			requested_mode = -1;
-			if (DEBUG)
-				printf("freedv_open: Failure because mode 2020 requires Avx support\n");
-			return 0;	// failure
-		}
-		if (handle_index >= 3) {	// Quisk provided codec2
-			CloseFreedv();
-			requested_mode = -1;
-			if (DEBUG)
-				printf("freedv_open: Failure because mode 2020 requires a system installation of codec2\n");
-			return 0;	// failure
-		}
-	}
-	if (requested_mode == FREEDV_MODE_700E) {
-		if (handle_index >= 3) {	// Quisk provided codec2
-			CloseFreedv();
-			requested_mode = -1;
-			if (DEBUG)
-				printf("freedv_open: Failure because mode 700E requires a system installation of codec2\n");
-			return 0;	// failure
-		}
-	}
-
-   	if ((requested_mode == FREEDV_MODE_700D || requested_mode == FREEDV_MODE_700E) && freedv_open_advanced) {
-       	struct freedv_advanced adv;
-       	adv.interleave_frames = interleave_frames;
-       	hF = freedv_open_advanced(requested_mode, &adv);
-   	}
-	else {
-        	hF = freedv_open(requested_mode);
-	}
-	if (hF == NULL) {
-		CloseFreedv();
-		requested_mode = -1;
-		return 0;	// failure
-	}
-	rx_channel[0].hFreedv = hF;
-	quisk_dvoice_freedv(&quisk_freedv_rx, &quisk_freedv_tx);
-	if (quisk_tx_msg[0])
-		freedv_set_callback_txt(hF, &put_next_rx_char, &get_next_tx_char, NULL);
-	else
-		freedv_set_callback_txt(hF, &put_next_rx_char, NULL, NULL);
-	if (freedv_set_callback_protocol)
- 		freedv_set_callback_protocol(hF, NULL, NULL, NULL);
-	if (freedv_set_callback_data)
-		freedv_set_callback_data(hF, NULL, my_datatx, NULL);
-	freedv_set_squelch_en(hF, quisk_freedv_squelch);
-	if (freedv_set_tx_bpf)
-		freedv_set_tx_bpf(hF, quisk_set_tx_bpf);
-	n_max_modem_samples = freedv_get_n_max_modem_samples(hF);
-	n_speech_sample_rate = 8000;
-	if (freedv_version >= 12 && freedv_get_speech_sample_rate)
-		n_speech_sample_rate = freedv_get_speech_sample_rate(hF);
-	n_modem_sample_rate = 8000;
-	if (freedv_version >= 11 && freedv_get_modem_sample_rate)
-		n_modem_sample_rate = freedv_get_modem_sample_rate(hF);
-	for (i = 0; i < MAX_RECEIVERS; i++) {
-		rx_channel[i].rxdata_index = 0;
-		rx_channel[i].speech_available = 0;
-		rx_channel[i].playing = 0;
-		if (rx_channel[i].demod_in)
-			free(rx_channel[i].demod_in);
-		rx_channel[i].demod_in = (COMP *)malloc(sizeof(COMP) * n_max_modem_samples);
-		if( rx_channel[i].demod_in == NULL) {
-			CloseFreedv();
-			requested_mode = -1;
-			return 0; // failure
-		}
-		if (i > 0) {
-			rx_channel[i].hFreedv = freedv_open(requested_mode);
-			if (rx_channel[i].hFreedv)
-				freedv_set_squelch_en(rx_channel[i].hFreedv, quisk_freedv_squelch);
-		}
-	}
-	if (DEBUG) printf("n_nom_modem_samples %d\n", freedv_get_n_nom_modem_samples(rx_channel[0].hFreedv));
-	if (DEBUG) printf("n_speech_samples %d\n", freedv_get_n_speech_samples(rx_channel[0].hFreedv));
-	if (DEBUG) printf("n_max_modem_samples %d\n", n_max_modem_samples);
-	if (DEBUG) {
-	 	if (freedv_version >= 11) printf("modem_sample_rate %d\n", n_modem_sample_rate);
-		if (freedv_version >= 12) printf("speech_sample_rate %d\n", n_speech_sample_rate);
-	}
-	freedv_current_mode = requested_mode;
-	return 1;		// success
-}
-
-void quisk_check_freedv_mode(void)
-{	// see if we need to change the mode
-	if (requested_mode == freedv_current_mode)
-		return;
-	if (DEBUG) printf("Change in mode to %d\n", requested_mode);
-	CloseFreedv();
-	if (requested_mode >= 0)
-		OpenFreedv();
-	else
-		requested_mode = -1;
-}
-
-PyObject * quisk_freedv_open(PyObject * self, PyObject * args)	// Called from the GUI thread before freedv is open
-{
-	if (!PyArg_ParseTuple (args, ""))
-		return NULL;
-	return PyInt_FromLong(OpenFreedv());
-}
-
-PyObject * quisk_freedv_close(PyObject * self, PyObject * args)	// Called from the GUI thread.
-{
-	if (!PyArg_ParseTuple (args, ""))
-		return NULL;
-	requested_mode = -1;		// request close
-	Py_INCREF (Py_None);
-	return Py_None;
-}
-
-PyObject * quisk_freedv_set_options(PyObject * self, PyObject * args, PyObject * keywds)	// Called from the GUI thread.
-{  // Call with keyword arguments ONLY to change parameters.  Call before quisk_freedv_open() to set an initial mode.
-	int mode=-1;				// Call again to change the mode.
-	int bpf=-1;
-	char * ptMsg=NULL;
-	static char * kwlist[] = {"mode", "tx_msg", "DEBUG", "squelch", "interleave_frames", "set_tx_bpf", NULL} ;
-
-	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|isiiii", kwlist, &mode, &ptMsg, &DEBUG, &quisk_freedv_squelch, &interleave_frames, &bpf))
-		return NULL;
-	if (ptMsg)
-		strMcpy(quisk_tx_msg, ptMsg, TX_MSG_SIZE);
-	if (bpf != -1) {
-		quisk_set_tx_bpf = bpf;
-		if (freedv_set_tx_bpf && rx_channel[0].hFreedv)
-			freedv_set_tx_bpf(rx_channel[0].hFreedv, quisk_set_tx_bpf);
-	}
-	if (mode == -1) {	// mode was not entered
-		;
-	}
-	else if (mode == FREEDV_MODE_700E && freedv_version >= 14) {
-		requested_mode = mode;
-	}
-	else if (mode == FREEDV_MODE_2020) {
-		if (checkAvxSupport() && handle_index <= 2 && freedv_version >= 13)
-			requested_mode = mode;
-	}
-	else if (freedv_version == 10) {
-		if (mode == 0)
-			requested_mode = mode;
-	}
-	else if (freedv_version == 11) {
-		if (mode <= 2)
-			requested_mode = mode;
-	}
-	else {
-		requested_mode = mode;
-	}
-	return PyInt_FromLong(requested_mode);	// Return the mode
-}
-
-PyObject * quisk_freedv_get_snr(PyObject * self, PyObject * args)	// Called from the GUI thread.
-{
-	float snr_est = 0.0;
-
-	if (!PyArg_ParseTuple (args, ""))
-		return NULL;
-	if (rx_channel[0].hFreedv)
-		freedv_get_modem_stats(rx_channel[0].hFreedv, NULL, &snr_est);
-	return PyFloat_FromDouble(snr_est);
-}
-
-PyObject * quisk_freedv_get_version(PyObject * self, PyObject * args)	// Called from the GUI thread.
-{
-	if (!PyArg_ParseTuple (args, ""))
-		return NULL;
-	if ( ! hLib)
-		GetAddrs();		// Get the entry points for funtions in the codec2 library
-	return PyInt_FromLong(freedv_version);
-}
-
-PyObject * quisk_freedv_set_squelch_en(PyObject * self, PyObject * args)	// Called from the GUI thread.
-{
-	int i,state;
-
-	if (!PyArg_ParseTuple (args, "i", &state))
-		return NULL;
-	quisk_freedv_squelch = state;
-	for (i = 0; i < MAX_RECEIVERS; i++) {
-		if( rx_channel[i].hFreedv )
-			freedv_set_squelch_en(rx_channel[i].hFreedv, state);
-	}
-	return PyInt_FromLong(state);
-}
-
-
+// This file is no longer used.

freedv.h

@@ -1,12 +1 @@
-extern int freedv_current_mode;         // the current FreeDV running mode
-
-#define FREEDV_MODE_1600        0
-#define FREEDV_MODE_700         1
-#define FREEDV_MODE_700B        2
-#define FREEDV_MODE_2400A       3
-#define FREEDV_MODE_2400B       4
-#define FREEDV_MODE_800XA       5
-#define FREEDV_MODE_700C        6
-#define FREEDV_MODE_700D        7
-#define FREEDV_MODE_2020        8
-#define FREEDV_MODE_700E        13
+// This header file is no longer used.

freedvpkg/README.txt

@@ -1,129 +1 @@
-
-Note:  The directory freedvpkg no longer contains source files.  It only contains copies of
-       the codec2 libraries for use by Quisk.  If any other files are present, delete them.
-
-FreeDV and Directory freedvpkg
-==============================
-
-FreeDV is the combination of the codec2 codec and the fdmdv modem.  It provides digital voice
-suitable for HF and VHF transmission.  Quisk has native (built-in) support for
-FreeDV.  Just push the FDV mode and talk.  This freedvpkg directory contains copies of the
-codec2 libraries for use by Quisk.
-
-You can also use the separate FreeDV program available at freedv.org, and the Quisk DGT-U mode
-which attaches to external digital programs. The setup is identical to fldigi and other external
-digital programs.
-
-Quisk will add the FDV button if specified on the Config/radio/Controls tab.
-If there is a problem with the freedv module, the button will be grayed out.
-
-The freedv module requires the codec2 library.  This library is included for Windows and for
-Ubuntu 18.04 LTS 32-bit and 64-bit.  For other systems (such as ARM) you will need to build another
-codec2.  Just try the FDV mode and see if it works.  It should always work on Windows, and may work
-on Linux.  If the FDV button is grayed out, you need a different codec2 than the one included.  Make
-a new codec2 library, and copy it to freedvpkg or to the correct system directory.
-
-2020 mode
-=========
-You can now use 2020 mode with Quisk.  Note however that 2020 mode requires a processor with AVX. 
-Also note a Raspberry PI (model 4B) does not have enough processing power even if neon is used
-and it is overclocked.
-
-There are two parts to this:
-1. build LPCNet
-2. rebuild codec2
-The details are found here: https://github.com/drowe67/codec2#freedv-2020-support-building-with-lpcnet
-
-Both liblpcnetfreedv and libcodec2 are needed.  If you are running under windows, the resultant
-liblpcnetfreedv.dll file needs to be placed in the quisk folder.  Libcodec2 may be placed in the
-quisk freedvpkg folder. Or install the DLL's in the correct system folders.
-
-Search Order
-============
-
-Quisk will search for a valid codec2 library in this order on Windows:
-1.  The system codec2 library installed outside of Quisk by another program.
-2.  freedvpkg/libcodec2.dll.     Not included. You can copy your own codec2 here but a system-wide installation is better.
-3.  freedvpkg/libcodec2_32.dll.  The 32-bit codec2 shipped with Quisk.
-4.  freedvpkg/libcodec2_64.dll.  The 64-bit codec2 shipped with Quisk.
-
-Quisk will search for a valid codec2 library in this order on Linux:
-1.  The system codec2 library installed outside of Quisk by another program.
-2.  freedvpkg/libcodec2.so.     Not included. You can copy your own codec2 here but a system-wide installation is better.
-3.  freedvpkg/libcodec2_32.so.  The 32-bit codec2 shipped with Quisk.
-4.  freedvpkg/libcodec2_64.so.  The 64-bit codec2 shipped with Quisk.
-
-How to Build a New codec2
-=========================
-
-The source for codec2 is available on github.  Or google for other sources or perhaps a pre-built library.
-If you need to compile codec2 from source, first change to a suitable directory (not the Quisk directory)
-and download the source with git:
-
-  git clone https://github.com/drowe67/codec2.git
-
-Then build codec2 using the directions found at https://github.com/drowe67/codec2/blob/master/INSTALL.
-The directions given below are current as of June 2020, but check for changes.
-Then copy the codec2 library to the freedvpkg directory under Quisk.
-
-Build a New codec2 on Linux
-===========================
-Create the codec2 shared library.  This assumes a 64-bit linux. Change the directory
-name from build_linux64 to build_linux32 for 32-bit linux.  Note the "../".
-
-  cd codec2
-  mkdir build_linux64
-  cd build_linux64
-  cmake -DCMAKE_BUILD_TYPE=Release ../
-  make
-  cd src
-  cp libcodec2.so my-quisk-directory/freedvpkg
-
-Build a New 32-bit codec2 on 64-bit Linux
-=========================================
-Make sure package libc6-dev-i386 is installed.
-Create the codec2 shared library.  Note the "../".
-
-  cd codec2
-  mkdir build_linux32
-  cd build_linux32
-  export CFLAGS=-m32
-  cmake -DCMAKE_BUILD_TYPE=Release ../
-  make
-  cd src
-  cp libcodec2.so my-quisk-directory/freedvpkg
-
-Build a New codec2 on Windows
-=============================
-For Windows you need to install MinGW-w64, MSYS2, and g++.  Use the mingw shell.  The Speex libraries
-are not needed by codec2, but are required for the Unit Test modules.  To build the Unit Test modules,
-you need to install Speex and add -DSPEEXDSP_INCLUDE_DIR=../speex/include/speex -DSPEEXDSP_LIBRARY=../speex/bin.
-
-  # Use mingw32.exe 32-bit shell:
-  cd codec2
-  mkdir build_win32
-  cd build_win32
-  cmake -G "MSYS Makefiles" -DCMAKE_BUILD_TYPE=Release -DUNITTEST=OFF  ../
-  make codec2
-  cd src
-  cp libcodec2.dll  my-quisk-directory/freedvpkg
-
-  # Use mingw64.exe 64-bit shell:
-  cd codec2
-  mkdir build_win64
-  cd build_win64
-  cmake -G "MSYS Makefiles" -DCMAKE_BUILD_TYPE=Release -DUNITTEST=OFF -DCMAKE_SYSTEM_PROCESSOR=x86_64 ../
-  make codec2
-  cd src
-  cp libcodec2.dll  my-quisk-directory/freedvpkg
-
-Testing
-=======
-You can just start Quisk and see if the FDV button is not grayed out, and FDV works.  Or you can
-test the import of freedv and look for error messages.
-
-  cd my-quisk-directory
-  python                                # Use the correct Python 2 or 3 in 32 or 64 bit
-    import _quisk
-    _quisk.freedv_get_version()         # This should return 13 or higher for a recent codec2
-
+This directory is no longer used by FreeDV. Please see the documentation.

handshake.c

@@ -0,0 +1,144 @@
+/*
+ * Copyright (C) 2016-2024  Davidson Francis <davidsondfgl@gmail.com>
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>
+ */
+#define _POSIX_C_SOURCE 200809L
+#include "base64.h"
+#include "sha1.h"
+#include "ws.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <strings.h>
+
+/**
+ * @dir src/
+ * @brief Handshake routines directory
+ *
+ * @file handshake.c
+ * @brief Handshake routines.
+ */
+
+/**
+ * @brief Gets the field Sec-WebSocket-Accept on response, by
+ * an previously informed key.
+ *
+ * @param wsKey Sec-WebSocket-Key
+ * @param dest source to be stored the value.
+ *
+ * @return Returns 0 if success and a negative number
+ * otherwise.
+ *
+ * @attention This is part of the internal API and is documented just
+ * for completeness.
+ */
+int get_handshake_accept(char *wsKey, unsigned char **dest)
+{
+	unsigned char hash[SHA1HashSize]; /* SHA-1 Hash.                   */
+	SHA1Context ctx;                  /* SHA-1 Context.                */
+	char *str;                        /* WebSocket key + magic string. */
+
+	/* Invalid key. */
+	if (!wsKey)
+		return (-1);
+
+	str = calloc(1, sizeof(char) * (WS_KEY_LEN + WS_MS_LEN + 1));
+	if (!str)
+		return (-1);
+
+	strncpy(str, wsKey, WS_KEY_LEN);
+	strcat(str, MAGIC_STRING);
+
+	SHA1Reset(&ctx);
+	SHA1Input(&ctx, (const uint8_t *)str, WS_KEYMS_LEN);
+	SHA1Result(&ctx, hash);
+
+	*dest = base64_encode(hash, SHA1HashSize, NULL);
+	*(*dest + strlen((const char *)*dest) - 1) = '\0';
+	free(str);
+	return (0);
+}
+
+/**
+ * @brief Finds the ocorrence of @p needle in @p haystack, case
+ * insensitive.
+ *
+ * @param haystack Target string to be searched.
+ * @param needle   Substring to search for.
+ *
+ * @returns If found, returns a pointer at the beginning of the
+ * found substring. Otherwise, returns NULL.
+ */
+static const char *strstricase(const char *haystack, const char *needle)
+{
+	size_t length;
+	for (length = strlen(needle); *haystack; haystack++)
+		if (!strncasecmp(haystack, needle, length))
+			return haystack;
+	return (NULL);
+}
+
+/**
+ * @brief Gets the complete response to accomplish a succesfully
+ * handshake.
+ *
+ * @param hsrequest  Client request.
+ * @param hsresponse Server response.
+ *
+ * @return Returns 0 if success and a negative number
+ * otherwise.
+ *
+ * @attention This is part of the internal API and is documented just
+ * for completeness.
+ */
+int get_handshake_response(char *hsrequest, char **hsresponse)
+{
+	unsigned char *accept; /* Accept message.     */
+	char *saveptr;         /* strtok_r() pointer. */
+	char *s;               /* Current string.     */
+	int ret;               /* Return value.       */
+
+	saveptr = NULL;
+	for (s = strtok_r(hsrequest, "\r\n", &saveptr); s != NULL;
+		 s = strtok_r(NULL, "\r\n", &saveptr))
+	{
+		if (strstricase(s, WS_HS_REQ) != NULL)
+			break;
+	}
+
+	/* Ensure that we have a valid pointer. */
+	if (s == NULL)
+		return (-1);
+
+	saveptr = NULL;
+	s       = strtok_r(s, " ", &saveptr);
+	s       = strtok_r(NULL, " ", &saveptr);
+
+	ret = get_handshake_accept(s, &accept);
+	if (ret < 0)
+		return (ret);
+
+	*hsresponse = malloc(sizeof(char) * WS_HS_ACCLEN);
+	if (*hsresponse == NULL)
+		return (-1);
+
+	strcpy(*hsresponse, WS_HS_ACCEPT);
+	strcat(*hsresponse, (const char *)accept);
+	strcat(*hsresponse, "\r\n\r\n");
+
+	free(accept);
+	return (0);
+}

hermes/hpsdr_widgets.py

@@ -0,0 +1,157 @@
+# Please do not change this widgets module for Quisk.  Instead copy
+# it to your own quisk_widgets.py and make changes there.
+#
+# This module is used to add extra widgets to the QUISK screen.
+
+import wx, math
+
+class BottomWidgets:	# Add extra widgets to the bottom of the screen
+  def __init__(self, app, hardware, conf, frame, gbs, vertBox):
+    self.config = conf
+    self.hardware = hardware
+    self.application = app
+    self.start_row = app.widget_row		# The first available row
+    self.start_col = app.button_start_col	# The start of the button columns
+    self.num_rows_added = 1
+    self.btnPreamp = app.QuiskCheckbutton(frame, self.OnBtnPreamp, text='Preamp')
+    gbs.Add(self.btnPreamp, (self.start_row, self.start_col), flag=wx.EXPAND)
+    bw, bh = self.btnPreamp.GetMinSize()
+    init = app.hermes_atten_dB
+    self.sliderAtten = app.SliderBoxHH(frame, 'Atten %d dB', init, 0, 31, self.OnAtten, True)
+    self.sliderAtten.idName = "RfAtten"
+    app.midiControls["RfAtten"]	= (self.sliderAtten, self.OnAtten)
+    hardware.ChangeAtten(init)
+    gbs.Add(self.sliderAtten, (self.start_row, self.start_col + 2), (1, 6), flag=wx.EXPAND)
+    if conf.button_layout == "Small screen":
+      # Display four data items in a single window
+      self.text_temperature = app.QuiskText1(frame, '', bh)
+      self.text_pa_current = app.QuiskText1(frame, '', bh)
+      self.text_fwd_power = app.QuiskText1(frame, '', bh)
+      self.text_swr = app.QuiskText1(frame, '', bh)
+      self.text_data = self.text_temperature
+      self.text_pa_current.Hide()
+      self.text_fwd_power.Hide()
+      self.text_swr.Hide()
+      b = app.QuiskPushbutton(frame, self.OnTextDataMenu, '..')
+      szr = self.data_sizer = wx.BoxSizer(wx.HORIZONTAL)
+      szr.Add(self.text_data, 1, flag=wx.ALIGN_CENTER_VERTICAL)
+      szr.Add(b, 0, flag=wx.ALIGN_CENTER_VERTICAL)
+      gbs.Add(szr, (self.start_row, self.start_col + 10), (1, 2), flag=wx.EXPAND)
+      # Make a popup menu for the data window
+      self.text_data_menu = wx.Menu()
+      item = self.text_data_menu.Append(-1, 'Temperature')
+      app.Bind(wx.EVT_MENU, self.OnDataTemperature, item)
+      item = self.text_data_menu.Append(-1, 'PA Current')
+      app.Bind(wx.EVT_MENU, self.OnDataPaCurrent, item)
+      item = self.text_data_menu.Append(-1, 'Fwd Power')
+      app.Bind(wx.EVT_MENU, self.OnDataFwdPower, item)
+      item = self.text_data_menu.Append(-1, 'SWR')
+      app.Bind(wx.EVT_MENU, self.OnDataSwr, item)
+    else:
+      szr = wx.BoxSizer(wx.HORIZONTAL)
+      gbs.Add(szr, (self.start_row, self.start_col + 10), (1, 18), flag=wx.EXPAND)
+      text_temperature = wx.StaticText(frame, -1, ' Temp 100DC XX', style=wx.ST_NO_AUTORESIZE)
+      size = text_temperature.GetBestSize()
+      text_temperature.Destroy()
+      self.text_temperature = wx.StaticText(frame, -1, '', size=size, style=wx.ST_NO_AUTORESIZE)
+      self.text_pa_current = wx.StaticText(frame, -1, '', size=size, style=wx.ST_NO_AUTORESIZE)
+      self.text_fwd_power = wx.StaticText(frame, -1, '', size=size, style=wx.ST_NO_AUTORESIZE)
+      self.text_swr = wx.StaticText(frame, -1, '', size=size, style=wx.ST_NO_AUTORESIZE)
+      flag=wx.ALIGN_LEFT|wx.ALIGN_CENTER_VERTICAL
+      szr.Add(self.text_temperature, 0, flag=flag)
+      szr.Add(self.text_pa_current, 0, flag=flag)
+      szr.Add(self.text_fwd_power, 0, flag=flag)
+      szr.Add(self.text_swr, 0, flag=flag)
+  def OnBtnPreamp(self, event):
+    self.hardware.ChangePreamp(self.btnPreamp.GetValue())
+  def OnAtten(self, event):
+    value = self.sliderAtten.GetValue()
+    self.hardware.ChangeAtten(value)
+    self.application.hermes_atten_dB = value
+  def Code2FwdRevWatts(self, fwd, rev):	# Convert the fwd/rev power code to watts forward and reverse
+    #print (self.hardware.hermes_rev_power, self.hardware.hermes_fwd_power)
+    fwd = self.hardware.InterpolatePower(fwd)
+    rev = self.hardware.InterpolatePower(rev)
+    # Which voltage is forward and reverse depends on the polarity of the current sense transformer
+    if fwd >= rev:
+      return fwd, rev
+    else:
+      return rev, fwd
+  def UpdateText(self):		# Called at intervals
+    conf = self.config
+    # Temperature
+    temp = self.hardware.hermes_temperature
+    if conf.calibrate_temp_20 == conf.calibrate_temp_40:
+      temp = " Temp ADC %4d" % temp
+    else:
+      m = 20.0 / (conf.calibrate_temp_40 - conf.calibrate_temp_20)
+      b = 20.0 - m * conf.calibrate_temp_20
+      temp = m * temp + b
+      temp = (" Temp %3.0f" % temp) + u'\u2103'
+    self.text_temperature.SetLabel(temp)
+    # power amp current
+    current = self.hardware.hermes_pa_current
+    if conf.calibrate_current_0 == conf.calibrate_current_1:
+      current = " Amps ADC %4d" % current
+    else:
+      m = 1.0 / (conf.calibrate_current_1 - conf.calibrate_current_0)
+      b = - m * conf.calibrate_current_0
+      current = m * current + b
+      current = " PA %4.0f ma" % (1000*current)
+    self.text_pa_current.SetLabel(current)
+    # forward and reverse peak power
+    if not self.hardware.power_interpolator:	# For no interpolator, show ADC
+      fwd = self.hardware.hermes_fwd_power
+      rev = self.hardware.hermes_rev_power
+      self.text_fwd_power.SetLabel("Fwd ADC %4d" % fwd)
+      self.text_swr.SetLabel("Rev ADC %4d" % rev)
+      return
+    fwd, rev = self.Code2FwdRevWatts(self.hardware.hermes_fwd_peak, self.hardware.hermes_rev_peak)
+    # forward less reverse power
+    power = fwd - rev
+    if power < 0.0:
+      power = 0.0
+    text = " PEP %3.1f watts" % power
+    self.text_fwd_power.SetLabel(text)
+    # SWR based on average power
+    fwd, rev = self.Code2FwdRevWatts(self.hardware.hermes_fwd_power, self.hardware.hermes_rev_power)
+    if fwd >= 0.05:
+      gamma = math.sqrt(rev / fwd)
+      if gamma < 0.98:
+        swr = (1.0 + gamma) / (1.0 - gamma)
+      else:
+        swr = 99.0
+      if swr < 9.95:
+        text = " SWR %4.2f" % swr
+      else:
+        text = " SWR %4.0f" % swr
+    else:
+      text = " SWR  ---"
+    self.text_swr.SetLabel(text)
+  def OnTextDataMenu(self, event):
+    btn = event.GetEventObject()
+    btn.PopupMenu(self.text_data_menu, (0,0))
+  def OnDataTemperature(self, event):
+    self.data_sizer.Replace(self.text_data, self.text_temperature)
+    self.text_data.Hide()
+    self.text_data = self.text_temperature
+    self.text_data.Show()
+    self.data_sizer.Layout()
+  def OnDataPaCurrent(self, event):
+    self.data_sizer.Replace(self.text_data, self.text_pa_current)
+    self.text_data.Hide()
+    self.text_data = self.text_pa_current
+    self.text_data.Show()
+    self.data_sizer.Layout()
+  def OnDataFwdPower(self, event):
+    self.data_sizer.Replace(self.text_data, self.text_fwd_power)
+    self.text_data.Hide()
+    self.text_data = self.text_fwd_power
+    self.text_data.Show()
+    self.data_sizer.Layout()
+  def OnDataSwr(self, event):
+    self.data_sizer.Replace(self.text_data, self.text_swr)
+    self.text_data.Hide()
+    self.text_data = self.text_swr
+    self.text_data.Show()
+    self.data_sizer.Layout()

hermes/quisk_hardware.py

@@ -15,6 +15,7 @@ from quisk_hardware_model import Hardware as BaseHardware
 
 DEBUG = 0
 DEBUG_I2C = 0
+DEBUG_SENSORS = 0
 
 class IOBoard:
   "This class controls the N2ADR IO Board for the HermesLite 2"
@@ -46,6 +47,8 @@ class IOBoard:
     self.old_receive = None
     self.slow = 0
   def HeartBeat(self):	# Called at 10 Hz for housekeeping tasks
+    if not self.hardware.is_HermesLite2:
+      return
     if not QS.get_params('rx_udp_started'):
       return
     if self.have_IO_Board is None:
@@ -154,6 +157,7 @@ class Hardware(BaseHardware):
     self.hermes_ip = ""
     self.hermes_code_version = -1
     self.hermes_board_id = -1
+    self.is_HermesLite2 = False
     self.hermes_temperature = 0.0
     self.hermes_fwd_power = 0.0
     self.hermes_rev_power = 0.0
@@ -166,7 +170,10 @@ class Hardware(BaseHardware):
     self.vna_count = 0
     self.vna_started = False
     self.repeater_freq = None		# original repeater output frequency
-    self.antenna_labels = ('Ant 0', 'Ant 1')	# labels for the antenna button
+    if conf.hardware_file_type == "Red Pitaya":
+      self.antenna_labels = ('Ant 0', 'Ant 1')	# labels for the antenna button
+    else:
+      self.antenna_labels = ('Ant 0', 'Ant 1', 'Ant 2', 'Ant 3')
     self.antenna_index = 0			# index of antenna in use
     self.delay_config = True			# Delay sending message to HL2 until after sound starts
     self.TFRC_counter = 0		# Call for power etc. at intervals
@@ -271,6 +278,7 @@ class Hardware(BaseHardware):
             self.hermes_ip = addr[0]
             self.hermes_code_version = data[9]
             self.hermes_board_id = data[10]
+            self.is_HermesLite2 = self.hermes_board_id == 6 and self.conf.hardware_file_type == "Hermes"
             QS.set_hermes_id(data[9], data[10])
             if data[0x16] >> 6 == 0:
               QS.set_params(bandscopeScale = 2048)
@@ -317,6 +325,7 @@ class Hardware(BaseHardware):
       bid = 6
       self.hermes_code_version = code
       self.hermes_board_id = bid
+      self.is_HermesLite2 = self.hermes_board_id == 6 and self.conf.hardware_file_type == "Hermes"
       QS.set_hermes_id(code, bid)
       st = 'Capture from Hermes device at specified IP %s' % self.hermes_ip
       found = True
@@ -331,9 +340,13 @@ class Hardware(BaseHardware):
   def open(self):
     self.delay_config = True	# Delay sending message to HL2 until after sound starts
     # This list only changes control bits; no use of WriteQueue()
-    for name in ('keyupDelay',
-        'hermes_lowpwr_tr_enable', 'hermes_PWM', 'hermes_disable_sync', 'hermes_power_amp', 'Hermes_BandDictEnTx'):
-      self.ImmediateChange(name)
+    self.ImmediateChange('keyupDelay')
+    self.ImmediateChange('Hermes_BandDictEnTx')
+    if self.is_HermesLite2:
+      for name in ('hermes_lowpwr_tr_enable', 'hermes_PWM', 'hermes_disable_sync', 'hermes_power_amp'):
+        self.ImmediateChange(name)
+    #else:	# DEBUG: Turn on power amp for testing OpenHPSDR radio with my Hermes Lite2
+    #  self.SetControlBit(0x09, 19, 1)
     return self.config_text
   def GetValue(self, name):	# return values stored in the hardware
     if name == 'Hware_Hl2_EepromIP':
@@ -492,7 +505,15 @@ class Hardware(BaseHardware):
     self.key_was_down = key_down
     if self.TFRC_counter >= 3:
       self.TFRC_counter = 0
-      self.hermes_temperature, self.hermes_fwd_power, self.hermes_rev_power, self.hermes_pa_current, self.hermes_fwd_peak, self.hermes_rev_peak = QS.get_hermes_TFRC()
+      AIN1_avg, AIN1_peak, AIN2_avg, AIN2_peak, AIN3, AIN4, AIN5, AIN6 = QS.get_hermes_TFRC()
+      if self.hermes_board_id == 6:
+        (self.hermes_fwd_power, self.hermes_fwd_peak, self.hermes_rev_power, self.hermes_rev_peak,
+         self.hermes_pa_current, self.hermes_temperature) = AIN1_avg, AIN1_peak, AIN2_avg, AIN2_peak, AIN3, AIN5
+      else:
+        (self.hermes_fwd_power, self.hermes_fwd_peak, self.hermes_rev_power, self.hermes_rev_peak,
+         self.hermes_pa_current, self.hermes_temperature) = AIN1_avg, AIN1_peak, AIN2_avg, AIN2_peak, AIN3, AIN5
+      if DEBUG_SENSORS:
+        print("AIN 1-6 %4d %4d %4d %4d %4d %4d" % (AIN1_avg, AIN2_avg, AIN3, AIN4, AIN5, AIN6))
       if self.application.bottom_widgets:
         self.application.bottom_widgets.UpdateText()
     if self.delay_config and QS.get_params('rx_udp_started'):
@@ -529,7 +550,9 @@ class Hardware(BaseHardware):
   def ChangeBandFilters(self):
     if not hasattr(self.application, "multi_rx_screen"):
       return	# Needed for the VNA program
-    self.SetControlBit(0x00, 13, self.antenna_index)
+    self.SetControlBit(0x00, 13, self.antenna_index & 0x01)
+    self.SetControlBit(0x00, 14, self.antenna_index & 0x02)
+    self.SetControlBit(0x00, 15, self.antenna_index > 0)
     highest = self.band
     freq = self.conf.BandEdge.get(highest, (0, 0))[0]
     for pane in self.application.multi_rx_screen.receiver_list:
@@ -606,12 +629,7 @@ class Hardware(BaseHardware):
     return (48000, 384000)
   ## Hardware AGC is no longer supported in HL2 identifying as version >=40   
   def ChangeAGC(self, value):
-    if value:
-      self.pc2hermes[2] |= 0x10		# C0 index == 0, C3[4]: AGC enable
-    else:
-      self.pc2hermes[2] &= ~0x10
-    QS.pc_to_hermes(self.pc2hermes)
-    if DEBUG: print ("Change AGC to", value)
+    pass
   ## Simpler LNA setting for HL2 identifying as version >=40, see HL2 wiki for details
   def ChangeLNA(self, value):		# LNA for Rx
     # value is -12 to +48
@@ -624,7 +642,7 @@ class Hardware(BaseHardware):
         value = 51 - value
     else:
       value = ((value+12) & 0x3f) | 0x40
-    self.pc2hermes[4 * 10 + 3] = value			# C0 index == 0x1010, C4[4:0] LNA 0-32 dB gain
+    self.pc2hermes[4 * 10 + 3] = value			# C0 index == 0b1010, C4[4:0] LNA 0-32 dB gain
     QS.pc_to_hermes(self.pc2hermes)
     if DEBUG: print ("Change LNA to", value)
   def ChangeTxLNA(self, value):		# LNA for Tx
@@ -634,15 +652,25 @@ class Hardware(BaseHardware):
     elif value > 48:
       value = 48
     value = ((value+12) & 0x3f) | 0x40 | 0x80
-    self.SetControlByte(0x0e, 3, value, False)		# C0 index == 0x0e, C3
+    self.SetControlByte(0x0e, 3, value, False)		# C0 index == 0b1110 == 0x0e, C3
     QS.pc_to_hermes(self.pc2hermes)
     if DEBUG: print ("Change Tx LNA to", value)
+  def ChangeAtten(self, value):
+    # value is 0 to 31
+    self.pc2hermes[4 * 10 + 3] = value & 0x1F			# C0 index == 0b1010, C4[4:0] attenuator, 0 to 31 dB
+    self.SetControlBit(0b1010, 5, value)			# enable attenuator
+    QS.pc_to_hermes(self.pc2hermes)
+    if DEBUG: print ("Change attenuator to", value)
+  def ChangePreamp(self, value):
+    self.SetControlBit(0b0000, 10, value)			# enable preamp
   def SetTxLevel(self):
     try:
       tx_level = self.conf.tx_level[self.band]
     except KeyError:
       tx_level = self.conf.tx_level.get(None, 127)	# The default
-    if self.mode[0:3] in ('DGT', 'FDV'):			# Digital modes; change power by a percentage
+    if not self.mode:
+      reduc = self.application.tx_level
+    elif self.mode[0:3] in ('DGT', 'FDV'):			# Digital modes; change power by a percentage
       reduc = self.application.digital_tx_level
     else:
       reduc = self.application.tx_level
@@ -651,7 +679,7 @@ class Hardware(BaseHardware):
       tx_level = 0
     elif tx_level > 255:
       tx_level = 255
-    self.pc2hermes[4 * 9] = tx_level			# C0 index == 0x1001, C1[7:0] Tx level
+    self.pc2hermes[4 * 9] = tx_level			# C0 index == 0b1001, C1[7:0] Tx level
     QS.pc_to_hermes(self.pc2hermes)
     if DEBUG: print("Change tx_level to", tx_level)
   def MultiRxCount(self, count):	# count == number of additional receivers besides the Tx/Rx receiver: 1, 2, 3
@@ -874,6 +902,8 @@ class Hardware(BaseHardware):
       QS.set_hermeslite_writepointer(0)
     return False
   def WriteQueue(self, wait=False):
+    if not self.is_HermesLite2:
+      return False
     self._wait_queue()
     # Send next write
     self.old_writequeue = self.pc2hermeslitewritequeue[:]

hermes/quisk_widgets.py

@@ -8,6 +8,7 @@ from __future__ import absolute_import
 from __future__ import division
 
 import math, wx, time
+import _quisk as QS
 from configure import ComboCtrl
 
 
@@ -24,8 +25,14 @@ class BottomWidgets:	# Add extra widgets to the bottom of the screen
   def Widgets_0x06(self, app, hardware, conf, frame, gbs, vertBox):
     self.num_rows_added = 1
     start_row = self.start_row
+    self.btnPS = app.QuiskCheckbutton(frame, self.OnBtnPS, text='PS')
+    self.btnPS.Enable(False)
+    gbs.Add(self.btnPS, (start_row, self.start_col), flag=wx.EXPAND)
+    self.btnPsCal = app.QuiskPushbutton(frame, self.OnBtnPsCal, 'PS Cal')
+    self.btnPsCal.active = False
+    gbs.Add(self.btnPsCal, (start_row, self.start_col + 1), flag=wx.EXPAND)
     self.atu_ctrl = ComboCtrl(frame, "ATU", ["Tune", "Bypass"], True)
-    gbs.Add(self.atu_ctrl, (start_row, self.start_col), (1, 2), flag=wx.EXPAND)
+    gbs.Add(self.atu_ctrl, (start_row, self.start_col + 2), (1, 2), flag=wx.EXPAND)
     bw, bh = self.atu_ctrl.GetMinSize()
     frame.Bind(wx.EVT_COMBOBOX_CLOSEUP, self.OnAtu)
     init = app.hermes_LNA_dB
@@ -33,7 +40,7 @@ class BottomWidgets:	# Add extra widgets to the bottom of the screen
     self.sliderLNA.idName = "RfLna"
     app.midiControls["RfLna"]	= (self.sliderLNA,	self.OnLNA)
     hardware.ChangeLNA(init)
-    gbs.Add(self.sliderLNA, (start_row, self.start_col + 2), (1, 8), flag=wx.EXPAND)
+    gbs.Add(self.sliderLNA, (start_row, self.start_col + 4), (1, 6), flag=wx.EXPAND)
     if conf.button_layout == "Small screen":
       # Display four data items in a single window
       self.text_temperature = app.QuiskText1(frame, '', bh)
@@ -93,6 +100,20 @@ class BottomWidgets:	# Add extra widgets to the bottom of the screen
     value = self.sliderLNA.GetValue()
     self.hardware.ChangeLNA(value)
     self.application.hermes_LNA_dB = value
+  def OnBtnPS(self, event):
+    if self.btnPS.GetValue():
+      QS.set_tx_audio(PsEnable=1)
+      self.btnPsCal.Enable(False)
+    else:
+      QS.set_tx_audio(PsEnable=0)
+      self.btnPsCal.Enable(True)
+  def OnBtnPsCal(self, event):
+    if len(self.application.multi_rx_screen.receiver_list) == 0:
+      self.application.multi_rx_screen.OnAddReceiver(event=None, hide_it=True)
+    self.btnPS.Enable(False)
+    self.btnPsCal.active = True
+    self.btnPsCal.SetColor(self.config.color_check_btn)
+    QS.set_tx_audio(PsCal=1)
   def Code2Temp(self):		# Convert the HermesLite temperature code to the temperature
     temp = self.hardware.hermes_temperature
     # For best accuracy, 3.26 should be a user's measured 3.3V supply voltage.
@@ -117,7 +138,13 @@ class BottomWidgets:	# Add extra widgets to the bottom of the screen
       return fwd, rev
     else:
       return rev, fwd
-  def UpdateText(self):
+  def UpdateText(self):		# Called at intervals
+    # Button state
+    if self.btnPsCal.active:
+      if not QS.get_tx_audio("PsCal"):
+        self.btnPsCal.active = False
+        self.btnPsCal.SetColorNormal()
+        self.btnPS.Enable(True)
     # Temperature
     temp = self.Code2Temp()
     temp = (" Temp %3.0f" % temp) + u'\u2103'
@@ -127,6 +154,12 @@ class BottomWidgets:	# Add extra widgets to the bottom of the screen
     current = " PA %4.0f ma" % (1000*current)
     self.text_pa_current.SetLabel(current)
     # forward and reverse peak power
+    if not self.hardware.power_interpolator:	# For no interpolator, show ADC
+      fwd = self.hardware.hermes_fwd_power
+      rev = self.hardware.hermes_rev_power
+      self.text_fwd_power.SetLabel("Fwd ADC %4d" % fwd)
+      self.text_swr.SetLabel("Rev ADC %4d" % rev)
+      return
     fwd, rev = self.Code2FwdRevWatts(self.hardware.hermes_fwd_peak, self.hardware.hermes_rev_peak)
     # forward less reverse power
     power = fwd - rev

hiqsdr/quisk_hardware.py

@@ -402,6 +402,8 @@ class Hardware(BaseHardware):
     if do_tx:
       try:
         self.rx_udp_socket.send(s)
+        if DEBUG:
+          self.PrintStatus(' Send', self.want_udp_status)
       except:
         pass
   def SetVNA(self, key_down=None, vna_start=None, vna_stop=None, vna_count=None, do_tx=False):

icons/quisk.svg

@@ -0,0 +1,221 @@
+<?xml version="1.0" encoding="UTF-8" standalone="no"?>
+<!-- Created with Inkscape (http://www.inkscape.org/) -->
+
+<svg
+   width="512"
+   height="512"
+   viewBox="0 0 512 512"
+   version="1.1"
+   id="svg5"
+   sodipodi:docname="quisk.svg"
+   inkscape:version="1.2.2 (b0a8486541, 2022-12-01)"
+   xmlns:inkscape="http://www.inkscape.org/namespaces/inkscape"
+   xmlns:sodipodi="http://sodipodi.sourceforge.net/DTD/sodipodi-0.dtd"
+   xmlns:xlink="http://www.w3.org/1999/xlink"
+   xmlns="http://www.w3.org/2000/svg"
+   xmlns:svg="http://www.w3.org/2000/svg">
+  <sodipodi:namedview
+     id="namedview7"
+     pagecolor="#ffffff"
+     bordercolor="#000000"
+     borderopacity="0.25"
+     inkscape:showpageshadow="2"
+     inkscape:pageopacity="0.0"
+     inkscape:pagecheckerboard="0"
+     inkscape:deskcolor="#d1d1d1"
+     inkscape:document-units="px"
+     showgrid="false"
+     inkscape:zoom="1"
+     inkscape:cx="36.5"
+     inkscape:cy="265"
+     inkscape:window-width="1854"
+     inkscape:window-height="1011"
+     inkscape:window-x="0"
+     inkscape:window-y="0"
+     inkscape:window-maximized="1"
+     inkscape:current-layer="layer1" />
+  <defs
+     id="defs2" />
+  <g
+     inkscape:label="Layer 1"
+     inkscape:groupmode="layer"
+     id="layer1">
+    <rect
+       style="fill:#c0c276;fill-opacity:1;stroke:#7180ff;stroke-width:48.671;stroke-dasharray:none;stroke-opacity:1"
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+       height="349.28082"
+       x="62.823105"
+       y="39.435188"
+       ry="63.709782" />
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+  </g>
+</svg>

import_quisk_api.c

@@ -32,6 +32,8 @@ with this:
 */
 
 #include <Python.h>
+#include <string.h>
+#include <stdio.h>
 
 void ** Quisk_API;		// array of pointers to functions and variables from module _quisk
 struct sound_conf * pt_quisk_sound_state;		// pointer to quisk_sound_state

is_key_down.c

@@ -1,5 +1,6 @@
 #include <Python.h>	// used by quisk.h
 #include <complex.h>	// Used by quisk.h
+#include <string.h>
 #include "quisk.h"
 
 // This module provides methods to access the state of the key.
@@ -168,6 +169,7 @@ static void modem_status(void)
 
 #include <fcntl.h>
 #include <sys/ioctl.h>
+#include <sys/time.h>
 #include <termios.h>
 
 static int fdComm = -1;			// File descriptor to read the serial port

makefil.py

@@ -0,0 +1,70 @@
+# This provides helper functions for our makefile.
+
+# Changes for MacOS support thanks to Mario, DL3LSM.
+# Changes for building from macports provided by Eric, KM4DSJ
+# Updated code for a Mac build contributed by Christoph, DL1YCF, December 2020.
+# Converted from setup.py to makefile by Jim Ahlstrom, June 2026
+
+import sys, os, subprocess, sysconfig
+
+if len(sys.argv) < 2:
+  sys.exit(2)
+
+job = sys.argv[1]
+
+def flags ():
+  # Fetch compiler flags and link flags from Python
+  result = subprocess.run(["python3-config", "--cflags"], capture_output=True, text=True, check=True)
+  cflags = result.stdout.strip()
+  cflags += " -fPIC"
+  result = subprocess.run(["python3-config", "--ldflags"], capture_output=True, text=True, check=True)
+  ldflags = result.stdout.strip()
+  ldflags += " -shared -lm -Wl,-O1 -Wl,-Bsymbolic-functions -Wl,-z,relro -g -fwrapv -O2 -lfftw3"
+  if sys.platform == "darwin":	# Build for Macintosh
+    cflags += " -DQUISK_HAVE_PORTAUDIO "
+    ldflags += " -lportaudio "
+    if os.path.isdir('/opt/local/include'):	# MacPorts
+      base_dir = '/opt/local'
+    elif os.path.isdir('/usr/local/include'):	# HomeBrew on macOS Intel 
+      base_dir = '/usr/local'
+    elif os.path.isdir('/opt/homebrew/include'): # HomeBrew on Apple Silicon
+      base_dir = '/opt/homebrew'
+    else:						# Regular build?
+      base_dir = '/usr'
+    if os.path.isfile(base_dir + "/include/pulse/pulseaudio.h"):
+      cflags += " -DQUISK_HAVE_PULSEAUDIO "
+      ldflags += " -lpulse "
+    cflags += " -I. -I%s/include " % base_dir
+    ldflags += " -L. -L%s/lib " % base_dir
+  elif "freebsd" in sys.platform:	#Build for FreeBSD
+    cflags += " -DQUISK_HAVE_PULSEAUDIO -I. -I/usr/local/include"
+    ldflags += " -lpulse -L. -L/usr/local/lib"
+  else:		# Linux
+    cflags += " -DQUISK_HAVE_ALSA -DQUISK_HAVE_PULSEAUDIO"
+    ldflags += " -lasound -lpulse"
+    if os.path.isfile("/usr/include/portaudio.h"):
+      cflags += " -DQUISK_HAVE_PORTAUDIO"
+      ldflags += " -lportaudio"
+  return cflags, ldflags
+
+if job == "--soapy":
+  # See if soapy is available
+  if os.path.isdir("/usr/include/SoapySDR") or os.path.isdir("/usr/local/include/SoapySDR"):
+    print('1')
+  else:
+    print('0')
+
+elif job == "--suffix":
+  # Determine the platform-specific extension suffix (e.g., .cpython-311-x86_64-linux-gnu.so)
+  print(sysconfig.get_config_var('EXT_SUFFIX'))
+
+elif job == "--cflags":
+  cc, ld = flags()
+  print(cc)
+
+elif job == "--ldflags":
+  cc, ld = flags()
+  print(ld)
+
+else:
+  sys.exit(1)

makefile

@@ -1,18 +1,93 @@
-.PHONY: quisk
+# This makefile replaces the old Python "setup.py build_ext" which has been deprecated.
+# See "quisk-old" below to make Quisk using the old setup.py.
 
-quisk:
+MODULE_NAME := _quisk
+OBJDIR := build/local
+SUBDIRS := afedrinet
+
+ifeq ($(shell python3 makefil.py --soapy), 1)
+SUBDIRS := $(SUBDIRS) soapypkg
+endif
+
+SRCS := \
+base64.c \
+extdemod.c \
+filter.c \
+freedv.c \
+handshake.c \
+is_key_down.c \
+microphone.c \
+quisk.c \
+quisk_wdsp.c \
+sha1.c \
+sound.c\
+sound_alsa.c \
+sound_directx.c \
+sound_portaudio.c \
+sound_pulseaudio.c \
+sound_wasapi.c \
+tci.c \
+utf8.c \
+utility.c \
+ws.c
+
+CFLAGS			:= $(shell python3 makefil.py --cflags)
+LDFLAGS			:= $(shell python3 makefil.py --ldflags)
+EXTENSION_SUFFIX	:= $(shell python3 makefil.py --suffix)
+TARGET := $(MODULE_NAME)$(EXTENSION_SUFFIX)
+
+#$(info Info CFLAGS $(CFLAGS))
+#$(info Info LDFLAGS $(LDFLAGS))
+#$(info Info TARGET $(TARGET))
+#$(info Info SUBDIRS $(SUBDIRS))
+
+OBJS := $(patsubst %.c, $(OBJDIR)/%.o, $(SRCS))
+
+.PHONY: all clean test
+.PHONY: $(SUBDIRS)
+
+# Default target
+all: $(OBJDIR) $(OBJDIR)/import_quisk_api.o ac2yd/remote.o $(TARGET) $(SUBDIRS)
+
+$(SUBDIRS):
+	$(MAKE) -C $@
+
+$(OBJDIR):
+	@mkdir -p $(OBJDIR)
+
+# Link the shared library object
+$(TARGET): $(OBJS)
+	$(CC) -o $@ $(OBJS) ac2yd/remote.o $(LDFLAGS)
+
+# Define the location for dependency files
+DEPS = $(patsubst %.o,$(OBJDIR)/%.d,$(notdir $(OBJS)))
+DEPFLAGS = -MMD -MP -MF $(OBJDIR)/$(notdir $*).d
+-include $(DEPS)
+
+# Compile the C source files into object files
+ac2yd/remote.o:	ac2yd/remote.c quisk.h filter.h
+	$(CC) -o $@ -c $< $(CFLAGS) $(DEPFLAGS)
+
+$(OBJDIR)/import_quisk_api.o: import_quisk_api.c
+	$(CC) -o $@ -c $< $(CFLAGS) $(DEPFLAGS)
+
+$(OBJDIR)/%.o: %.c | $(OBJDIR)
+	$(CC) -o $@ -c $< $(CFLAGS) $(DEPFLAGS)
+
+clean:
+	-rm -f  $(OBJDIR)/*.o $(OBJDIR)/*.d
+	-rm -f	ac2yd/*.o
+	-rm -f	$(MODULE_NAME)*.so
+	for dir in $(SUBDIRS); do \
+		$(MAKE) -C $$dir clean; \
+	done
+
+# Run a test script to verify that the module can be imported
+test: $(TARGET)
+	@python3 -c "import $(MODULE_NAME); print('Successfully imported:', $(MODULE_NAME).__file__)"
+
+
+
+.PHONY: quisk-old
+quisk-old:
 	python3 setup.py build_ext --force --inplace
-	@echo
-
-quisk3:
-	python3 setup.py build_ext --force --inplace
-	@echo
-
-soapy3:
-	(cd soapypkg; make soapy3)
-
-afedrinet3:
-	(cd afedrinet; make afedrinet3)
-
-perseus3:
-	(cd perseuspkg; make perseus3)

microntrxpkg/__init__.py

@@ -0,0 +1 @@
+#

microntrxpkg/quisk_hardware.py

@@ -0,0 +1,418 @@
+# This provides support for the SDR Micron TRX by David Fainitski, N7DDC
+
+# On Windows this uses Python module ftd2xx. Install with pip.
+# On Linux it uses ftdi1. Install python3-ftdi1 with your package manager.
+
+import sys, wx, traceback, time
+from quisk_hardware_model import Hardware as BaseHardware
+
+DEBUG = 1	# print debug messages during operation
+WRITE_LOG = 1	# write the data received from the MicronTRX during self.log_time to the file self.log_name
+#WRITE_LOG = 2	# receive data from the file self.log_name instead of from USB
+
+# https://github.com/Dfinitski/SDR-Micron
+
+# Short Control Packet type "SCP" from PC to Micron TRX, 16 bytes, MSB first (big endian)
+#
+# (0 - 7) Preamble, seven 0x55 and 0xD5
+# (8, 9, 10) "SCP"
+#(11) RX0 enable, 0 - 1
+#(12) RX1 enable, 0 - 1
+#(13) BS0 enable, 0 - 1
+#(14) TX0 enable, 0 - 1
+#(15) CDC enable, 0 - 1
+
+# Universal control packet type "UCP" from PC to TRX or from TRX to PC, 64 bytes, MSB first (big-endian)
+#
+# (0 - 7) Preamble, seven 0x55 and 0xD5
+# (8, 9, 10) "UCP"
+# (11) read only FW prefix MSB
+# (12) read only FW prefix 
+# (13) read only FW prefix LSB
+# (14) read only char '-'
+# (15) Not used
+# (16) Not used
+# (17) Not used
+# (18, 19, 20, 21) - 4 bytes RX0 frequency
+# (22, 23, 24, 25) - 4 bytes RX1 frequency
+# (26, 27, 28, 29) - 4 bytes BS0 frequency
+# (30, 31, 32, 33) - 4 bytes TX0 frequency
+# (34) - Attenuator, 0 - 31 (dB value)
+# (35) - BS0 Rate
+#          0 - 48 kHz, 
+#          1 - 96 kHz,
+#          2 - 192 kHz,
+#          3 - 384 kHz,
+#          4 - 768 kHz,
+#          5 - 1536 kHz
+# (36) LNA enable
+# (37) TX level 0 - 255
+# (38) RX BPF bypass 0 - 1
+# (39) BS0 mode
+#         0 - 4096 buffered samples
+#         1 - 8192 buffered samples
+#         2 - 16384 buffered samples
+# (40) Not used
+# (41) Not used
+# (42) read-only PTT state from device 0-1
+# (43) Mic Boost, 0 - 1
+# (44) TX LPF Bypass 0 - 1
+# (45) PTT to device 1 - 0
+# (46) CW mode on 0 - 1
+# (47) CW key manipulation from PC to device 1 - 0
+# (48) Read only PWR, Output power in Watts
+# (49) Read only SWR x.x, from 1.0 to 9.9  as a natural number without decimal point
+# (50) Key Speed 10 - 63 ppm (20 by default)
+# (51) Paddle swap 0, 1
+# (52) Key mode, 0 = straight, 01 = Mode A, 10 = Mode B
+# (53) Key weight, 33 to 66, (50 by default)
+# (54) Key auto spacing, 0, 1
+# (55) ST Level, 0 - 255 sight tone level
+# (56) ST Freq in tens Hertz, 40 - 100 (400 - 1000Hz), 70 by default 
+# (57) Not used
+# (58) Delay ON, 8 - 255 ms (25 by default) - time for TX carrier delay is needed for transmitter preparing to TX
+# (59) Hang out time in tens ms 0 - 255 (0 to 2550 msec) (250 ms by default) needed to keep transmitter ready after PTT
+# (60) Read only CW key manipulation from device to PC
+# (61) Read only FW2 - firmware version MSB
+# (62) Read only FW1
+# (63) Read only FW0 - firmware version LSB
+
+# Define the name of the hardware and the items on the hardware screen (see quisk_conf_defaults.py):
+################ Receivers SdrMicronTRX, The SDR Micron TRX project by David Fainitski.
+## hardware_file_name		Hardware file path, rfile
+# This is the file that contains the control logic for each radio.
+#hardware_file_name = 'microntrxpkg/quisk_hardware.py'
+
+
+class Hardware(BaseHardware):
+  sample_rates =  [48]  #[48, 96, 192, 384, 768, 1536]
+  BS0_samples = {0:4096, 1:8192, 2:16384}
+  BS0_bytes = {0:4096 * 2, 1:8192 * 2, 2:16384 * 2}	# Two bytes per sample
+  def __init__(self, app, conf):
+    BaseHardware.__init__(self, app, conf)
+    self.usb_windows = None
+    self.usb_linux = None
+    self.prefix = b'\x55\x55\x55\x55\x55\x55\x55\xD5'
+    self.read_data = bytearray()
+    self.tx_samples = b''
+    self.rf_gain_labels = ('RF 0', 'RF -10', 'RF -20', 'RF -30')
+    self.index = 1
+    self.old_vfo = 0
+    self.sdrmicron_clock = 76800000
+    self.sdrmicron_decim = 1600
+    self.bscope_data = bytearray()
+    self.have_version = False
+    self.frame_msg = ''
+    self.remaining_BS0 = -1
+    self.log_name = "microntrxpkg/microntrx_log.dat"
+    self.log_time = 5.0		# seconds of log data
+    self.log_time0 = None	# starting time.time() to start log file
+    self.log_fp = None
+    self.pc_control = bytearray(64)
+    self.pc_control[0:8] = self.prefix
+    self.pc_control[8:11] = b'UCP'
+    self.got_defaults = 0
+    self.pc_enables = bytearray(16)
+    self.pc_enables[0:8] = self.prefix
+    self.pc_enables[8:11] = b'SCP'
+    self.fw_version = bytearray
+    
+    if conf.fft_size_multiplier == 0:
+      conf.fft_size_multiplier = 3		# Set size needed by VarDecim
+
+  def open(self):	# This method must return a string showing whether the open succeeded or failed.
+    if WRITE_LOG == 1:
+      self.log_fp = open(self.log_name, "wb")
+    elif WRITE_LOG == 2:
+      self.log_fp = open(self.log_name, "rb")
+    rx_bytes = 3	# rx_bytes is the number of bytes in each I or Q sample: 1, 2, 3, or 4
+    rx_endian = 1	# rx_endian is the order of bytes in the sample array: 0 == little endian; 1 == big endian
+    self.InitSamples(rx_bytes, rx_endian)	# Initialize: read samples from this hardware file and send them to Quisk
+    self.InitTxSamples(2, 1, 48000)			# Initialize Tx samples to send to the hardware
+    bs_bytes = 2
+    bs_endian = 1
+    #self.InitBscope(bs_bytes, bs_endian, self.sdrmicron_clock, self.BS1_samples[self.pc_control[40]])	# Initialize bandscope
+    if WRITE_LOG == 2:
+      return "Read data from %s" % self.log_name
+    elif sys.platform == 'win32':
+      return self.open_win()
+    else:
+      return self.open_linux()
+
+  def open_win(self):
+    try:
+      import ftd2xx as d2xx
+    except:
+      return 'The ftd2xx module is missing'
+    enum = d2xx.createDeviceInfoList() # quantity of FTDI devices
+    if enum == 0:
+        return 'There are no FTDI devices available'
+    for i in range(enum):  # Searching and opening needed device
+      a = d2xx.getDeviceInfoDetail(i)
+      if DEBUG:
+        print ("%d: description %s, serial %s" % (i, a['description'], a['serial']))
+      if a['description']==b'SDR-Micron':
+        try:
+          usb = d2xx.openEx(a['serial'])
+        except:
+          return 'Device SDR-Micron could not be opened'
+        usb.setBitMode(255, 0)  # reset
+        time.sleep(0.1)
+        usb.setBitMode(255, 64)  # Configure FT2232H into Sync FIFO mode 
+        usb.setTimeouts(100, 100) # read, write
+        usb.setLatencyTimer(2)
+        usb.setUSBParameters(32, 32) # in_tx_size, out_tx_size
+        time.sleep(1.5) # waiting for device initialisation
+        usb.read(usb.getQueueStatus()) # clean the usb data buffer
+        self.frame_msg = a['description'].decode('utf-8') + '   S/N - ' + a['serial'].decode('utf-8')
+        self.usb_windows = usb
+        return self.frame_msg
+    return 'Device SDR-Micron was not found'
+
+  def open_linux(self):		# Not finished
+    try:
+      import ftdi1
+    except:
+      return 'The ftdi1 module is missing'
+    usb = ftdi1.new()
+    # Open Channel A (interface 1 in libftdi)
+    ftdi1.set_interface(usb, ftdi1.INTERFACE_A)
+    rc = ftdi1.usb_open(usb, 0x0403, 0x6014) # 0x6014 is the standard FT232H PID
+    if rc == 0:
+      ftdi1.usb_reset(usb)
+      ftdi1.set_usb_read_timout(usb, 1)
+      ftdi1.set_usb_write_timout(usb, 5)
+      ftdi1.set_latency_timer(usb, 2)
+      # Enable Sync FIFO mode
+      # 0xFF = pin mask, 0x40 = BITMODE_SYNCFF
+      ftdi1.set_bitmode(usb, 0xFF, ftdi1.BITMODE_SYNCFF) 
+      if DEBUG:
+        print("Sync FIFO active via ftdi1.")
+      self.usb_linux = usb
+      return "Device SDR-Micron opened"
+    return 'Device SDR-Micron was not found'
+
+  def usb_read(self):
+    if WRITE_LOG == 2:
+      data = self.log_fp.read(120)
+      return data
+    elif self.usb_windows:
+      count = self.usb_windows.getQueueStatus()	# Read all available bytes
+      if count > 0:
+        return self.usb_windows.read(count)	# return Python bytes
+    elif self.usb_linux:
+      count, data = ftdi1.read_data(self.usb_linux, 1024)
+      if count > 0:
+        return data	# data is Python bytes
+      elif count < 0 and DEBUG:
+        print("Error %d in usb_read", count)
+    return None
+
+  def usb_write(self, data):
+    data = bytes(data)        # Convert bytearray to bytes
+    if self.usb_windows:
+      try:
+        self.usb_windows.write(data)
+      except:
+        if DEBUG:
+          print('Error while usb_write')  
+    elif self.usb_linux:
+      count = ftdi1.write_data(self.usb_linux, data)
+      if count != len(data) and DEBUG:
+        print("Failure in usb_write_data %d %d", count, len(data))
+
+  def close(self):
+    if self.usb_windows:
+      self.usb_windows.setBitMode(255, 0)  # reset
+      time.sleep(0.5)
+      self.usb_windows.close()
+      self.usb_windows = None
+    elif self.usb_linux:
+      ftdi1.usb_close(self.usb_linux)
+      ftdi1.free(self.usb_linux)
+      self.usb_linux = None
+    
+  def OnButtonRfGain(self, event):
+    btn = event.GetEventObject()
+    n = btn.index
+    if n == 0:
+      self.pc_control[34] = 0
+    elif n == 1:
+      self.pc_control[34] = 10
+    elif n == 2:
+      self.pc_control[34] = 20
+    elif n == 3:
+      self.pc_control[34] = 30
+    if DEBUG:
+      print ('pc_control[34]', self.pc_control[34])
+    self.rx_control_upd()
+
+  def ChangeFrequency(self, tx_freq, vfo, source='', band='', event=None):
+    if DEBUG:
+      print('ChangeFrequency tx_freq', tx_freq, 'vfo', vfo)
+    if tx_freq and tx_freq > 0:
+      tx_freq = int(tx_freq - self.transverter_offset)
+      self.pc_control[30] = self.pc_control[18] = (tx_freq >> 24) & 0xFF	# MSB
+      self.pc_control[31] = self.pc_control[19] = (tx_freq >> 16) & 0xFF
+      self.pc_control[32] = self.pc_control[20] = (tx_freq >>  8) & 0xFF
+      self.pc_control[33] = self.pc_control[21] = (tx_freq      ) & 0xFF	#LSB
+      self.rx_control_upd()
+    if vfo:
+      vfo =  int(vfo - self.transverter_offset)
+      if vfo != self.old_vfo:
+        self.old_vfo = vfo
+        self.pc_control[18] = (vfo >> 24) & 0xFF	# MSB
+        self.pc_control[19] = (vfo >> 16) & 0xFF
+        self.pc_control[20] = (vfo >>  8) & 0xFF
+        self.pc_control[21] = (vfo      ) & 0xFF	#LSB
+        self.rx_control_upd()
+    return tx_freq, vfo
+
+  def OnSpot(self, level):
+    pass
+
+  def OnChangeRxTx(self, is_rx):
+    self.pc_control[45] = is_rx
+    self.rx_control_upd()
+  
+  def VarDecimGetChoices(self): # Return a list/tuple of strings for the decimation control.
+    return list(map(str, self.sample_rates)) # convert integer to string
+
+  def VarDecimGetLabel(self):		# return a text label for the control
+    return "Sample rate ksps"
+  
+  def VarDecimGetIndex(self):		# return the current index
+    return self.index
+  
+  def VarDecimSet(self, index=None): # return sample rate
+    if index is None: # initial call to set the sample rate before the call to open()
+      rate = self.application.vardecim_set
+      try:
+        self.index = self.sample_rates.index(rate // 1000)
+      except:
+        self.index = 0
+    else:
+      self.index = index
+    self.pc_control[35] = self.index
+    rate = self.sample_rates[self.index] * 1000
+    if DEBUG:
+      print("varDecimSet index", self.index, "rate", rate)
+    self.rx_control_upd()
+    return rate
+  
+  def VarDecimRange(self):  # Return the lowest and highest sample rate.
+    #return (48000, 1536000)
+    return (48000, 48000)
+
+  def StartSamples(self):	# called by the sound thread
+    self.pc_enables[11] = 1	# RX0 enable
+    #self.pc_enables[13] = 1	# BS0 enable
+    self.usb_write(self.pc_enables)
+    if DEBUG:
+      print ("Start samples")
+  
+  def StopSamples(self):	# called by the sound thread
+    self.pc_enables[11] = 0
+    self.pc_enables[13] = 0
+    self.usb_write(self.pc_enables)
+    if DEBUG:
+      print ("Stop Samples")
+
+  def rx_control_upd(self):
+    if self.got_defaults:
+      self.usb_write(self.pc_control)
+
+  def GetRxSamples(self):	# Read from the MicronTRX. Called frequently from the sound thread.
+    # Send Tx samples to the hardware
+    self.tx_samples += self.GetTxSamples()
+    while len(self.tx_samples) >= 492:
+      data = self.prefix + b'TX0\x00\x00\x00\x00\x00' + self.tx_samples[0:492]
+      self.usb_write(data)
+      #print(data[0:24], len(data))
+      #i = data[16] << 8 | data[17]
+      #print ("%8d" % i)
+      self.tx_samples = self.tx_samples[492:]
+    # Read messages from USB
+    data = self.usb_read()
+    if data:
+      self.read_data += data
+      if DEBUG > 1:
+        print("Read %s bytes, buffer length %d" % (len(data), len(self.read_data)))
+      if WRITE_LOG == 1:
+        if self.log_fp:
+          if self.log_time0 is None:
+            self.log_time0 = time.time()
+          if time.time() - self.log_time0 < self.log_time:
+            self.log_fp.write(data)
+          else:
+            self.log_fp.close()
+            self.log_fp = None
+    while True:
+      start = self.read_data.find(self.prefix)
+      if start < 0:	# No prefix
+        return
+      if start > 0:
+        del self.read_data[0:start]
+        if DEBUG:
+          print ("USB blocks lost sync")
+      if len(self.read_data) < 11:
+        return
+      block_type = self.read_data[8:11]
+      if block_type == b'BS0':
+        if len(self.read_data) < 508:
+          return
+        packet_number = self.read_data[11]
+        if packet_number == 0:			# start of a block of data
+          self.remaining_BS0 = self.BS0_bytes[self.pc_control[39]]
+          self.bscope_data = self.read_data[16:508]		# 492 bytes
+          self.remaining_BS0 -= 492
+        elif self.remaining_BS0 >= 492:
+          self.bscope_data += self.read_data[16:508]		# 492 bytes
+          self.remaining_BS0 -= 492
+        elif self.remaining_BS0 <= 0:
+          pass
+        else:
+          self.bscope_data += self.read_data[16:self.remaining_BS0 + 16] 
+          self.remaining_BS0 = -1
+          # self.AddBscopeSamples(self.bscope_data)
+        if DEBUG > 1:
+          print("BS0 buffer length %d" % len(self.bscope_data))
+        del self.read_data[0:508]
+      elif block_type == b'CDM':
+        if len(self.read_data) < 180:
+          return
+        if DEBUG > 1:
+          print("CDM")
+        del self.read_data[0:180]
+      elif block_type == b'RX0':
+        if len(self.read_data) < 508:
+          return
+        if self.read_data[12]:
+          self.GotClip()
+        self.AddRxSamples(self.read_data[16:508])
+        if DEBUG > 1:
+          print("RX0")
+        del self.read_data[0:508]
+      elif block_type == b'RX1':
+        if len(self.read_data) < 508:
+          return
+        if DEBUG > 1:
+          print("RX1")
+        del self.read_data[0:508]
+      elif block_type == b'UCP':
+        if len(self.read_data) < 64:
+          return
+        if self.have_version is False:
+          self.have_version = True
+          self.fw_version = self.read_data[11:15] + self.read_data[61:64]
+          self.frame_msg += '   F/W version - ' + self.fw_version.decode('utf-8')
+          self.application.main_frame.SetConfigText(self.frame_msg)
+        self.pc_control[11:64] = self.read_data[11:64] # save UCP content with defaults
+        self.got_defaults = 1;
+        if DEBUG > 1:
+          print("UCP")
+        del self.read_data[0:64]
+      else:
+        del self.read_data[0:11] 
+        if DEBUG:
+          print("Unknown USB block type")

microphone.c

@@ -3,12 +3,12 @@
 #include <math.h>
 #include <sys/time.h>
 #include <complex.h>
+#include <string.h>
 #include <fftw3.h>
 #include "quisk.h"
 #include <sys/types.h>
 #include "microphone.h"
 #include "filter.h"
-#include "freedv.h"
 
 #ifdef MS_WINDOWS
 #include <winsock2.h>
@@ -73,8 +73,10 @@ static double modulation_index = 1.6;	// For FM transmit, the modulation index
 static int is_vox = 0;					// Is the VOX level exceeded?
 static int vox_level = CLIP16;			// VOX trigger level as a number 0 to CLIP16
 static int timeVOX = 2000;				// VOX hang time in milliseconds
-static int tx_sample_rate = 48000;	// Used for SoapySDR
+static int tx_sample_rate = 48000;	// Used for SoapySDR and Tx samples from Python
 static int reverse_tx_sideband;
+static int PsEnable = 0;
+static int PsCal = 0;
 
 static int doTxCorrect = 0;				// Corrections for UDP sample transmit
 static double TxCorrectLevel;
@@ -243,11 +245,9 @@ static void init_alc(struct alc * pt, int size)
 			case DGT_U:
 			case DGT_L:
 			case DGT_IQ:
-				pt->gain_now[i] = 1.4;
-				break;
 			case FDV_U:
 			case FDV_L:
-				pt->gain_now[i] = 2.0;
+				pt->gain_now[i] = 1.4;
 				break;
 			default:
 				pt->gain_now[i] = 1.0;
@@ -372,7 +372,7 @@ static void process_alc(complex double * cSamples, int count, struct alc * pt, r
 
 static int tx_filter(complex double * filtered, int count)
 {	// Input samples are creal(filtered), output is filtered.  The input rate must be 8000 or 48000 sps.
-	int i, is_ssb;
+	int i, is_ssb, mic_interp;
 	int sample_rate = 8000;
 	double dsample, dtmp, magn;
 	complex double csample;
@@ -423,9 +423,10 @@ static int tx_filter(complex double * filtered, int count)
 		}
 		return 0;
 	}
+	mic_interp = MIC_OUT_RATE / sample_rate;
 	// check size of dsamples[] and csamples[] buffer
 	if (count > samples_size) {
-		samples_size = count * 2;
+		samples_size = count * 2 * mic_interp;
 		if (dsamples)
 			free(dsamples);
 		if (csamples)
@@ -541,8 +542,8 @@ static int tx_filter(complex double * filtered, int count)
 		// remove clipping distortion
 		count = quisk_cDecimate(csamples, count, &cfiltAudio3, 1);
 		// Interpolate up to 48000
-		if (MIC_OUT_RATE != sample_rate)
-			count = quisk_cInterpolate(csamples, count, &cfiltInterp, MIC_OUT_RATE / sample_rate);
+		if (mic_interp != 1)
+			count = quisk_cInterpolate(csamples, count, &cfiltInterp, mic_interp);
 		// convert back to 16 bits
 		for (i = 0; i < count; i++) {
 			filtered[i] = csamples[i] * CLIP16;
@@ -573,8 +574,8 @@ static int tx_filter(complex double * filtered, int count)
 		// remove clipping distortion
 		count = quisk_dFilter(dsamples, count, &dfiltAudio3);
 		// Interpolate up to 48000
-		if (MIC_OUT_RATE != sample_rate)
-			count = quisk_dInterpolate(dsamples, count, &dfiltInterp, MIC_OUT_RATE / sample_rate);
+		if (mic_interp != 1)
+			count = quisk_dInterpolate(dsamples, count, &dfiltInterp, mic_interp);
 		// convert back to 16 bits
 		for (i = 0; i < count; i++) {
 			filtered[i] = dsamples[i] * CLIP16;
@@ -614,7 +615,7 @@ static int tx_filter_digital(complex double * filtered, int count)
 		quisk_filt_dInit(&filter1, quiskDgtFilt48Coefs, sizeof(quiskDgtFilt48Coefs)/sizeof(double));    // pass 1350, stop 1650
 	}
 	if ( ! filtered) {	// Change to rxMode
-		quisk_filt_tune(&filter1, 1650.0 / 48000, rxMode != DGT_L && rxMode != LSB);
+		quisk_filt_tune(&filter1, 1650.0 / 48000, rxMode != DGT_L && rxMode != LSB && rxMode != FDV_L);
 		return 0;
 	}
 	for (i = 0; i < count; i++)	// FIR bandpass filter; separate into I and Q
@@ -623,123 +624,6 @@ static int tx_filter_digital(complex double * filtered, int count)
 	return count;
 }
 
-static int tx_filter_freedv(complex double * filtered, int count, int encode)
-{	// Input samples are creal(filtered), output is filtered.
-	// This filter is used for digital voice.
-	// changes by Dave Roberts, G8KBB, for additional modes June, 2020.
-	int i;
-	// The input sample rate is mic_sample_rate, either 8000 or 48000 sps.
-	// The FreeDV codec requires an input n_speech_sample_rate of either 8000 or 16000 sps depending on the mode. Input is always real.
-	// The FreeDV codec will output samples at n_modem_sample_rate of 8000, 16000 or 48000 sps depending on the mode, and output may be real or complex.
-	double dtmp, magn, dsample;
-	static int samples_size = 0;
-	static int last_rx_mode;
-	static int last_freedv_mode;
-	static int do_init = 1;
-	static double aaa, bbb, ccc, Xmin, Xmax, Ymax;
-	static double time_long, time_short;
-	static double inMax=0.3;
-	static double * dsamples = NULL;
-	static struct quisk_cFilter filter2;
-	static struct quisk_dFilter filtDecim;
-	static struct quisk_cFilter cfiltInterp;
-
-	if (do_init) {		// initialization
-		//QuiskWavWriteOpen(&hWav, "jim.wav", 3, 1, 4, 8000, 1.0 / CLIP16);
-		do_init = 0;
-		last_rx_mode = -1;
-		last_freedv_mode = -1;
-		memset(&filter2, 0, sizeof(filter2));
-		quisk_filt_dInit(&filtDecim,  quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));   // pass 3000, stop 4000
-		quisk_filt_cInit(&cfiltInterp, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
-		Ymax = pow(10.0,  - 1 / 20.0);				// maximum y
-		Xmax = pow(10.0,  3 / 20.0);				// x where slope is zero; for x > Xmax, y == Ymax
-		Xmin = Ymax - fabs(Ymax - Xmax);			// x where slope is 1 and y = x; start of compression
-		//printf ("Xmin %f\n", Xmin);
-		aaa = 1.0 / (2.0 * (Xmin - Xmax));			// quadratic
-		bbb = -2.0 * aaa * Xmax;
-		ccc = Ymax - aaa * Xmax * Xmax - bbb * Xmax;
-	}
-	if (last_freedv_mode != freedv_current_mode) {		// correct filter2 depends on freedv mode
-		last_freedv_mode = freedv_current_mode;
-		last_rx_mode = -1;	// re-tune new filter
-		if (filter2.cSamples)
-			free(filter2.cSamples);
-		filter2.cSamples = NULL;
-		if (filter2.cpxCoefs)
-			free(filter2.cpxCoefs);
-		filter2.cpxCoefs = NULL;
-		if (filter2.cBuf)
-			free(filter2.cBuf);
-		filter2.cBuf = NULL;
-		switch(freedv_current_mode) {
-		case FREEDV_MODE_700D:		// filter2 is not used
-			break;
-		// TODO: add modes that return real samples that must be converted to complex
-		default:
-			quisk_filt_cInit(&filter2, quiskFilt53D2Coefs, sizeof(quiskFilt53D2Coefs)/sizeof(double));  // pass 1500, stop 1800
-			break;
-		}
-	}
-	if (last_rx_mode != (int)rxMode) {	// change to sideband
-		last_rx_mode = rxMode;
-		if (rxMode == FDV_U)	 // upper sideband
-			quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / n_modem_sample_rate, 1);
-		else if (rxMode == FDV_L)   // lower sideband
-			quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / n_modem_sample_rate, 0);
-	}
-	if ( ! filtered)
-		return 0;
-	// check size of dsamples[] buffer
-	if (count > samples_size) {
-		samples_size = count * 2;
-		if (dsamples)
-			free(dsamples);
-		dsamples = (double *)malloc(samples_size * sizeof(double));
-	}
-	// copy to dsamples[]
-	for (i = 0; i < count; i++)
-		dsamples[i] = creal(filtered[i]) / CLIP16;	// normalize to 1.0000
-	// Decimate to 8000 Hz or to 16000 depending on mode (and hence setting of sample rate)
-	if (quisk_sound_state.mic_sample_rate == 48000)
-		count = quisk_dDecimate(dsamples, count, &filtDecim, quisk_sound_state.mic_sample_rate / n_speech_sample_rate);
-	else if (quisk_sound_state.mic_sample_rate == 8000 && n_speech_sample_rate != 8000)
-		QuiskPrintf("Failure to convert input rate in tx_filter_freedv\n");
-	// Measure average peak input audio level and limit
-	dtmp = 1.0 / n_speech_sample_rate;		// sample time
-	time_long   = 1.0 - exp(- dtmp / 3.000);
-	time_short  = 1.0 - exp(- dtmp / 0.005);
-	for (i = 0; i < count; i++) {
-		dsample = dsamples[i];
-		magn = fabs(dsample);
-		if (magn > inMax)
-			inMax = inMax * (1 - time_short) + time_short * magn;
-		else if(magn > mic_agc_level)
-			inMax = inMax * (1 - time_long) + time_long * magn;
-		else
-			inMax = inMax * (1 - time_long) + time_long * mic_agc_level;
-		dsample = dsample / inMax * Xmin * 0.7;
-		magn = fabs(dsample);
-		if (magn < Xmin)
-			dsamples[i] = dsample;
-		else if (magn > Xmax)
-			dsamples[i] = copysign(Ymax, dsample);
-		else
-			dsamples[i] = copysign(aaa * magn * magn + bbb * magn + ccc, dsample);
-		dsamples[i] = dsamples[i] * CLIP16;
-	}
-	//QuiskWavWriteD(&hWav, dsamples, count);
-	if (encode && pt_quisk_freedv_tx)   // Encode audio into digital modulation
-		count = (* pt_quisk_freedv_tx)(filtered, dsamples, count);
-	//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
-	if (filter2.cSamples)	// convert float samples to complex with an analytic filter
-		count = quisk_cCDecimate(filtered, count, &filter2, 1);
-	// Interpolate up to 48000
-	if (MIC_OUT_RATE != n_modem_sample_rate ) 
-		count = quisk_cInterpolate(filtered, count, &cfiltInterp, MIC_OUT_RATE / n_modem_sample_rate);
-	return count;
-}
-
 PyObject * quisk_get_tx_filter(PyObject * self, PyObject * args)
 {  // return the TX filter response to display on the graph
 // This is for debugging.  Change quisk.py to call QS.get_tx_filter() instead
@@ -876,6 +760,8 @@ static void quisk_hermes_tx_add(complex double * cSamples, int tx_count, int key
 		if (hermes_read_index >= HERMES_TX_BUF_SHORTS)
 			hermes_read_index -= HERMES_TX_BUF_SHORTS;
 	}
+	// Send the transmit samples to the predistortion logic. Modify the samples in place.
+	PreDistort(NULL, NULL, 0, cSamples, tx_count);
 	hermes_num_samples += tx_count;
 	for (i = 0; i < tx_count; i++) {			// Put transmit mic samples into the buffer
 		hermes_buf[hermes_write_index++] = (short)cimag(cSamples[i]);
@@ -891,6 +777,7 @@ void quisk_hermes_tx_send(int tx_socket, int * tx_records)
 {	// Send one UDP block of mic samples using the Metis-Hermes protocol.  Timing is from blocks received, rate is 48k.
 	// If the key is up we send samples anyway, but the samples are zero.
 	int i, j, offset, sent, ratio;
+	int PsInUse;
 	short s;
 	unsigned char sendbuf[1032];
 	unsigned char * pt_buf;
@@ -898,6 +785,7 @@ void quisk_hermes_tx_send(int tx_socket, int * tx_records)
 	static unsigned char C0_index = 0;
 	complex double cw_samples[63 * 2];
 	static double writequeue_time0 = 0;
+	static unsigned char rx1_freq[4];	// frequency of the first receiver
 
 	//printf("hermes_tx_send start 1: hermes_num_samples %d\n", hermes_num_samples);
 	if (tx_records == NULL) {
@@ -910,6 +798,7 @@ void quisk_hermes_tx_send(int tx_socket, int * tx_records)
 		last_play_state = quisk_play_state;
 		serial_key_samples(NULL, 0);
 	}
+	PsInUse = PsEnable || PsCal;
 	//printf("hermes_tx_send start 2: hermes_num_samples %d\n", hermes_num_samples);
 	ratio = quisk_sound_state.sample_rate / 48000;		// send rate is 48 ksps
 	//printf ("quisk_hermes_tx_send ratio %d count %d\n", ratio, *tx_records);
@@ -953,15 +842,33 @@ void quisk_hermes_tx_send(int tx_socket, int * tx_records)
 	sendbuf[13] = quisk_pc_to_hermes[offset++];		// C2
 	sendbuf[14] = quisk_pc_to_hermes[offset++];		// C3
 	sendbuf[15] = quisk_pc_to_hermes[offset++];		// C4
-	if (C0_index == 0) {	// Do not change receiver count without stopping Hermes and restarting
+	switch (C0_index) {
+	case 0:	// Do not change receiver count without stopping Hermes and restarting
 		sendbuf[15] = quisk_multirx_count << 3 | 0x04;	// Send the old count, not the changed count
 		if (hermes_mox_bit)		// send filter selection on J16
 			sendbuf[13] = hermes_filter_tx << 1;
 		else
 			sendbuf[13] = hermes_filter_rx << 1;
+		break;
+	case 2:		// Rx1 frequency
+		rx1_freq[0] = sendbuf[12];
+		rx1_freq[1] = sendbuf[13];
+		rx1_freq[2] = sendbuf[14];
+		rx1_freq[3] = sendbuf[15];
+		break;
+	case 3:		// Rx2 frequency
+		if (PsInUse && hermes_mox_bit) {
+			sendbuf[12] = rx1_freq[0];
+			sendbuf[13] = rx1_freq[1];
+			sendbuf[14] = rx1_freq[2];
+			sendbuf[15] = rx1_freq[3];
+		}
+		break;
+	case 10:
+		if (PsInUse && hermes_mox_bit)
+			sendbuf[13] |= 0x40;	// assert pure signal
+		break;
 	}
-        else if ( ! quisk_is_vna && C0_index == 9) {
-        }
 	if (++C0_index > 16)
 		C0_index = 0;
 	pt_buf = sendbuf + 16;
@@ -1010,15 +917,33 @@ void quisk_hermes_tx_send(int tx_socket, int * tx_records)
 		sendbuf[525] = quisk_pc_to_hermes[offset++];		// C2
 		sendbuf[526] = quisk_pc_to_hermes[offset++];		// C3
 		sendbuf[527] = quisk_pc_to_hermes[offset++];		// C4
-		if (C0_index == 0) {
-			sendbuf[527] = quisk_multirx_count << 3 | 0x04;		// Send the old count, not the changed count
+		switch (C0_index) {
+		case 0:	// Do not change receiver count without stopping Hermes and restarting
+			sendbuf[527] = quisk_multirx_count << 3 | 0x04;	// Send the old count, not the changed count
 			if (hermes_mox_bit)		// send filter selection on J16
 				sendbuf[525] = hermes_filter_tx << 1;
 			else
 				sendbuf[525] = hermes_filter_rx << 1;
+			break;
+		case 2:		// Rx1 frequency
+			rx1_freq[0] = sendbuf[524];
+			rx1_freq[1] = sendbuf[525];
+			rx1_freq[2] = sendbuf[526];
+			rx1_freq[3] = sendbuf[527];
+			break;
+		case 3:		// Rx2 frequency
+			if (PsInUse && hermes_mox_bit) {
+				sendbuf[524] = rx1_freq[0];
+				sendbuf[525] = rx1_freq[1];
+				sendbuf[526] = rx1_freq[2];
+				sendbuf[527] = rx1_freq[3];
+			}
+			break;
+		case 10:
+			if (PsInUse && hermes_mox_bit)
+				sendbuf[525] |= 0x40;	// assert pure signal
+			break;
 		}
-                else if ( ! quisk_is_vna && C0_index == 9) {
-                }
 		if (++C0_index > 16)
 			C0_index = 0;
 	}
@@ -1084,12 +1009,47 @@ static void transmit_udp(complex double * cSamples, int count)
 
 static void transmit_mic_carrier(complex double * cSamples, int count, double level)
 {	// send a CW carrier instead of mic samples
+#if 0
+	// transmit a saw tooth
+	static int up = 1;
+	static int ramp = 0;
+	int i;
+	int max = 5000;
+	static int pause = 5000 / 10;
+	int half_max = (int)(0.1 / 0.2339 * max);
+	for (i = 0; i < count; i++) {
+		cSamples[i] = (double)ramp / max * CLIP16;
+		if (ramp == half_max && pause) {
+			pause--;
+		}
+		else if (up) {
+			if (ramp == max) {
+				up = 0;
+				pause = max / 10;
+			}
+			else {
+				ramp += 1;
+			}
+		}
+		else {
+			if (ramp == 0) {
+				up = 1;
+				pause = max / 10;
+			}
+			else {
+				ramp -= 1;
+			}
+		}
+		//printf("Ramp %d\n", ramp);
+	}
+#endif
 #if 1
 	// transmit a carrier equal to the number of samples
 	int i;
 	for (i = 0; i < count; i++)
 		cSamples[i] = level * CLIP16;
-#else
+#endif
+#if 0
 	// replace mic samples with a sin wave
 	int i;
 	double freq;
@@ -1260,6 +1220,10 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 			count *= interp;
 			transmit_mic_carrier(cSamples, count, quiskSpotLevel / 1000.0);
 		}
+		else if (tci_tx_audio_client) {
+			count = tx_filter_digital(cSamples, count);	// filter samples, minimal processing
+			process_alc(cSamples, count, &tx_alc, DGT_U);
+		}
 		else switch (rxMode) {
 		case LSB:		// LSB
 		case USB:		// USB
@@ -1301,6 +1265,8 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 		case DGT_L:
 		case DGT_IQ:
 		case DGT_FM:
+		case FDV_U:		// FreeDV
+		case FDV_L:
 			count = tx_filter_digital(cSamples, count);	// filter samples, minimal processing
 			process_alc(cSamples, count, &tx_alc, rxMode);
 			break;
@@ -1308,11 +1274,6 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 			count *= interp;
 			transmit_mic_imd(cSamples, count, quiskImdLevel / 1000.0);
 			break;
-		case FDV_U:	// FDV
-		case FDV_L:
-			count = tx_filter_freedv(cSamples, count, 1);
-			process_alc(cSamples, count, &tx_alc, rxMode);
-			break;
 		default:
 			break;
 		}
@@ -1344,7 +1305,7 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 		;	// hermes handles this itself
 	else if (quisk_play_state == SOFTWARE_CWKEY)
 		serial_key_samples(cSamples, count);
-        if(quisk_pt_sample_write) {	// Used for SoapySDR
+        if(quisk_pt_sample_write && key_down) {	// Used for SoapySDR and samples from Python
 		// Interpolate the mic samples to the Tx sample rate
 //printf("Tx sample rate %i\n", tx_sample_rate);
 		switch (tx_sample_rate) {
@@ -1403,11 +1364,6 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 			count = tx_filter(cSamples, count);
 			process_alc(cSamples, count, &tx_alc, rxMode);
 			break;
-		case FDV_U:	// FDV
-		case FDV_L:
-			count = tx_filter_freedv(cSamples, count, 0);
-			process_alc(cSamples, count, &tx_alc, rxMode);
-			break;
 		default:
 			for (i = 0; i < count; i++)
 				cSamples[i] = 0;
@@ -1429,14 +1385,31 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
 	return count;
 }
 
+PyObject * quisk_get_tx_audio(PyObject * self, PyObject * args)
+{
+	const char * name;
+
+	if (!PyArg_ParseTuple (args, "s", &name))
+		return NULL;
+
+	if (strcmp(name, "PsEnable") == 0)
+		return PyInt_FromLong(PsEnable);
+	if (strcmp(name, "PsCal") == 0)
+		return PyInt_FromLong(PsCal);
+	Py_INCREF (Py_None);
+	return Py_None;
+}
+
+
 PyObject * quisk_set_tx_audio(PyObject * self, PyObject * args, PyObject * keywds)
 {  /* Call with keyword arguments ONLY; change Tx audio parameters */
 	static char * kwlist[] = {"vox_level", "vox_time", "mic_clip", "mic_preemphasis", "tx_sample_rate",
-		"reverse_tx_sideband", NULL} ;
+		"reverse_tx_sideband", "PsEnable", "PsCal", NULL} ;
 	int vlevel = -9999, clevel = -9999;
 
-	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiidii", kwlist,
-			&vlevel, &timeVOX, &clevel, &quisk_mic_preemphasis, &tx_sample_rate, &reverse_tx_sideband))
+	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiidiiii", kwlist,
+			&vlevel, &timeVOX, &clevel, &quisk_mic_preemphasis, &tx_sample_rate, &reverse_tx_sideband,
+			&PsEnable, &PsCal))
 		return NULL;
 	if (vlevel != -9999)
 		vox_level = (int)(pow(10.0, vlevel / 20.0) * CLIP16);	// Convert dB to 16-bit sample
@@ -1560,7 +1533,6 @@ void quisk_set_tx_mode(void)	// called when the mode rxMode is changed
 	tx_filter(NULL, 0);
 	tx_filter_digital(NULL, 0);
 	transmit_udp(NULL, 0);
-	tx_filter_freedv(NULL, 0, 0);
 	serial_key_samples(NULL, 0);
 }
 
@@ -1606,3 +1578,527 @@ static void serial_key_samples(complex double * cSamples, int count)	// called f
 		cSamples[i] = ampl * CLIP16;
 	}
 }
+
+static void spline(double x[], double y[], int n, double yp1, double ypn, double y2[])
+{
+	int i,k;
+	double p,qn,sig,un,*u;
+
+	u = (double *)malloc(n * sizeof(double));
+	if (yp1 > 0.99e30)
+		y2[1]=u[1]=0.0;
+	else {
+		y2[1] = -0.5;
+		u[1]=(3.0/(x[2]-x[1]))*((y[2]-y[1])/(x[2]-x[1])-yp1);
+	}
+	for (i=2;i<=n-1;i++) {
+		sig=(x[i]-x[i-1])/(x[i+1]-x[i-1]);
+		p=sig*y2[i-1]+2.0;
+		y2[i]=(sig-1.0)/p;
+		u[i]=(y[i+1]-y[i])/(x[i+1]-x[i]) - (y[i]-y[i-1])/(x[i]-x[i-1]);
+		u[i]=(6.0*u[i]/(x[i+1]-x[i-1])-sig*u[i-1])/p;
+	}
+	if (ypn > 0.99e30)
+		qn=un=0.0;
+	else {
+		qn=0.5;
+		un=(3.0/(x[n]-x[n-1]))*(ypn-(y[n]-y[n-1])/(x[n]-x[n-1]));
+	}
+	y2[n]=(un-qn*u[n-1])/(qn*y2[n-1]+1.0);
+	for (k=n-1;k>=1;k--)
+		y2[k]=y2[k]*y2[k+1]+u[k];
+	free(u);
+}
+
+static void polint(double xa[], double ya[], double x, double *y)
+{
+	int i,m,ns=1;
+	double den,dif,dift,ho,hp,w, dy;
+	double c[4], d[4];
+
+	dif=fabs(x-xa[1]);
+	for (i=1;i<=3;i++) {
+		if ( (dift=fabs(x-xa[i])) < dif) {
+			ns=i;
+			dif=dift;
+		}
+		c[i]=ya[i];
+		d[i]=ya[i];
+	}
+	*y=ya[ns--];
+	for (m=1;m<3;m++) {
+		for (i=1;i<=3-m;i++) {
+			ho=xa[i]-x;
+			hp=xa[i+m]-x;
+			w=c[i+1]-d[i];
+			den = ho - hp;
+			if ( (den=ho-hp) == 0.0)
+				printf("Error in routine polint\n");
+			den=w/den;
+			d[i]=hp*den;
+			c[i]=ho*den;
+		}
+		*y += (dy=(2*ns < (3-m) ? c[ns+1] : d[ns--]));
+	}
+}
+
+static void splint(double xa[], double ya[], double y2a[], int n, double x, double *y)
+{
+	static int klo = 1;
+	int khi,k;
+	double h,b,a;
+
+	khi = klo + 1;
+	if (x < xa[klo] || x > xa[khi]) {
+		klo=1;
+		khi=n;
+		while (khi-klo > 1) {
+			k=(khi+klo) >> 1;
+			if (xa[k] > x) khi=k;
+			else klo=k;
+		}
+	}
+	h=xa[khi]-xa[klo];
+	if (h == 0.0)
+		printf("Bad xa input to routine splint\n");
+	a=(xa[khi]-x)/h;
+	b=(x-xa[klo])/h;
+	*y=a*ya[klo]+b*ya[khi]+((a*a*a-a)*y2a[klo]+(b*b*b-b)*y2a[khi])*(h*h)/6.0;
+}
+
+#define NUM_BINS	21
+#define BIN_SPACING	(1.0 / (NUM_BINS - 1))
+#define BIN_TOL		(BIN_SPACING * 0.20)
+#define BLOCK_MSECS	2000		// Time of one data block.
+#define MILLISECS(x)	(48 * (x))	// Convert milliseconds to samples at 48000 samples per second.
+#define DEBUG_PS	0
+#define HOLD		0		// Make calibration run permanent
+#define RX1_GAIN	0.238
+void PreDistort(complex double * amp_in_samples, complex double * amp_out_samples, int nSamples, complex double * tx_samples, int num_tx)
+{
+	int i, bin, num_bin_data;
+	double G0, phase, mag_in, mag_out, new_mag, aaa, ppp;
+	double error, xp, xpg, xp0, xpg0;
+	double quadX[4], quadY[4], quadZ[4];
+	complex double in_sample, out_sample;
+	static int state = 0;
+	static int Fault;
+	static int isCalibrated = 0;
+	static int ramp_counter, ramp_count0;
+	static int input_samples = 0;
+	static double tx_to_RX1_gain = 0;
+	static double max_out_mag, starting_phase;
+#if DEBUG_PS
+	static double clip_tx, clip_mag_in, clip_aaa;
+#endif
+#if DEBUG_PS > 2
+	static int old_state;
+	static double time0;
+#endif
+	static int spline_P_size;
+	static double spline_P_X[NUM_BINS + 1];
+	static double spline_P_mag[NUM_BINS + 1];
+	static double spline_P_mag2[NUM_BINS + 1];
+	static double spline_P_phase[NUM_BINS + 1];
+	static double spline_P_phase2[NUM_BINS + 1];
+
+	struct BPD {
+		int XP_count;
+		double table_XP;
+		double XP_mag;
+		double mag_X;
+		double mag_P;
+		double phase_P;
+		complex double XP_gain;
+		double XP_gain_mag;
+		double XP_gain_phase;
+	};
+	static struct BPD Bins[NUM_BINS];
+		
+	static enum {
+		NothingHappens,
+		SendCW,
+		StopCW
+	} tx_state = NothingHappens;
+
+	if (quisk_multirx_count == 0) {
+		state = 0;
+		return;
+	}
+	if (PsCal) {
+		if (state == 0) {
+			isCalibrated = 0;
+			state = 10;
+#if DEBUG_PS > 2
+			time0 = QuiskTimeSec();
+#endif
+		}
+	}
+	else if ( ! PsEnable) {
+		state = 0;
+		return;
+	}
+	else if ( ! hermes_mox_bit) {
+		state = 0;
+		return;
+	}
+
+	if (num_tx) {
+		input_samples += num_tx;
+		if (isCalibrated) {	// perform predistortion of Tx samples; multiply input X by P
+			for (i = 0; i < num_tx; i++) {
+				mag_in = cabs(tx_samples[i]) / CLIP16;
+				splint(spline_P_X, spline_P_mag, spline_P_mag2, spline_P_size, mag_in, &aaa);
+				splint(spline_P_X, spline_P_phase, spline_P_phase2, spline_P_size, mag_in, &ppp);
+				tx_samples[i] *= aaa * cexp(I * ppp);
+				new_mag = mag_in * aaa;
+				if (new_mag > 1.0) {
+					tx_samples[i] /= new_mag;
+#if DEBUG_PS
+					if (new_mag > 1.0001 && new_mag > clip_tx) {
+						clip_tx = new_mag;
+						clip_mag_in = mag_in;
+						clip_aaa = aaa;
+					}
+#endif
+				}
+			}
+		}
+		switch (tx_state) {
+		case NothingHappens:
+		default:
+			break;
+		case SendCW:	// Start sending a CW carrier.
+			for (i = 0; i < num_tx; i++) {
+				if (ramp_counter > 0)	// slow increase
+					tx_samples[i] = CLIP16 * (1.0 - (double)ramp_counter-- / ramp_count0);
+				else
+					tx_samples[i] = CLIP16;
+			}
+			break;
+		case StopCW:	// Stop sending a CW carrier.
+			for (i = 0; i < num_tx; i++) {
+				if (ramp_counter > 0)	// slow decrease
+					tx_samples[i] = (double)CLIP16 * ramp_counter-- / ramp_count0;
+				else
+					tx_samples[i] = 0;
+			}
+			break;
+		}
+	}
+
+	if (nSamples <= 0)
+		return;
+
+#if DEBUG_PS > 2
+	if (state != old_state) {
+		aaa = cabs(amp_in_samples[0] / CLIP32);
+		printf("State %2d time %6.3lf sample %12.6f\n", old_state, QuiskTimeSec() - time0, aaa);
+		old_state = state;
+		time0 = QuiskTimeSec();
+	}
+#endif
+
+	switch (state) {
+	case 0:		// Start a timer
+		tx_state = NothingHappens;
+		input_samples = 0;
+		starting_phase = 0;
+		if ( ! HOLD)
+			state++;
+		break;
+	case 1:
+		if (input_samples >= MILLISECS(200)) {
+#if DEBUG_PS
+			clip_tx = 0.0;
+#endif
+			for (bin = 0; bin < NUM_BINS; bin++) {		// initialize
+				Bins[bin].XP_count = 0;
+				Bins[bin].XP_mag = 0;
+				Bins[bin].XP_gain = 0;
+			}
+			input_samples = 0;
+			state++;
+		}
+		break;
+	case 2:
+	case 15:
+		if (input_samples >= MILLISECS(BLOCK_MSECS)) {
+			num_bin_data = 0;
+			Fault = 0;
+			for (bin = 0; bin < NUM_BINS; bin++) {
+				if (Bins[bin].XP_count) {
+					num_bin_data++;
+					Bins[bin].XP_mag /= Bins[bin].XP_count;
+					Bins[bin].XP_gain /= Bins[bin].XP_count;
+					if (starting_phase == 0 && bin >= NUM_BINS / 2) {
+						starting_phase = carg(Bins[bin].XP_gain);
+#if DEBUG_PS
+						printf("starting_phase %.3f\n", starting_phase);
+#endif
+					}
+				}
+			}
+			if (PsCal && num_bin_data < NUM_BINS - 4) {
+				printf("PS calibration failed to fill all bins ");
+				for (bin = 0; bin < NUM_BINS; bin++)
+					if (Bins[bin].XP_count)
+						printf("1");
+					else
+						printf("0");
+				printf("\n");
+				Fault = 1;
+			}
+			state++;
+		}
+		else {
+			for (i = 0; i < nSamples; i++) {
+				in_sample = amp_in_samples[i] / CLIP32 / RX1_GAIN;
+				mag_in = cabs(in_sample);
+				if (tx_to_RX1_gain < mag_in) {
+					tx_to_RX1_gain = mag_in;
+#if DEBUG_PS > 2
+					printf("tx RX1 gain %.6f, max_out_mag %.6f\n", tx_to_RX1_gain, max_out_mag);
+#endif
+				}
+				bin = (int)(mag_in / BIN_SPACING + 0.5);
+				if ( ! bin)
+					continue;
+				error = fabs(mag_in - Bins[bin].table_XP);
+				if (error < BIN_TOL) {
+					out_sample = amp_out_samples[i] / CLIP32;
+					Bins[bin].XP_gain += out_sample / in_sample;
+					Bins[bin].XP_mag += mag_in;
+					Bins[bin].XP_count++;
+				}
+			}
+			if (state == 15) {
+				if (tx_state != StopCW) {
+					ramp_counter = ramp_count0 = MILLISECS(BLOCK_MSECS - 500);
+					tx_state = StopCW;
+				}
+			}
+		}
+		break;
+	case 3:
+	case 16:
+#if DEBUG_PS
+		if (clip_tx != 0.0) {
+			printf ("clip output sample %.6f, mag_in %.6f, predistort %.6f\n", clip_tx, clip_mag_in, clip_aaa);
+			clip_tx = 0.0;
+		}
+#endif
+		for (bin = 1; bin < NUM_BINS - 1; bin++) {
+			if (Bins[bin].XP_count > 0) {	// This bin has data.
+				Bins[bin].XP_gain_mag = cabs(Bins[bin].XP_gain);
+				phase = carg(Bins[bin].XP_gain) - starting_phase;
+				if (phase < - M_PI)
+					phase += 2 * M_PI;
+				else if (phase > M_PI)
+					phase -= 2 * M_PI;
+				Bins[bin].XP_gain_phase = phase;
+			}
+			else {
+				Bins[bin].XP_gain_mag = 1.0;
+				Bins[bin].XP_gain_phase = 0.0;
+			}
+		}
+		Bins[0].XP_count = 1;
+		Bins[0].XP_mag = 0.0;
+		i = 1;
+		for (bin = 1; bin < 6; bin++) {
+			if (Bins[bin].XP_count) {
+				quadX[i] = Bins[bin].XP_mag;
+				quadY[i] = Bins[bin].XP_gain_mag;
+				quadZ[i] = Bins[bin].XP_gain_phase;
+				if (i == 3)
+					break;
+				i++;
+			}
+		}
+		if (i == 3) {
+			polint(quadX, quadY, 0.0, &Bins[0].XP_gain_mag);
+			polint(quadX, quadZ, 0.0, &Bins[0].XP_gain_phase);
+		}
+		else {
+			Bins[0].XP_gain_mag = 1.0;
+			Bins[0].XP_gain_phase = 0.0;
+			Fault = 1;
+			if (DEBUG_PS || PsCal)
+				printf("Failure to find G0 at zero\n");
+		}
+
+		Bins[NUM_BINS - 1].XP_count = 1;
+		Bins[NUM_BINS - 1].XP_mag = 1.0;
+		i = 3;
+		for (bin = NUM_BINS - 2; bin >= NUM_BINS - 6; bin--) {
+			if (Bins[bin].XP_count) {
+				quadX[i] = Bins[bin].XP_mag;
+				quadY[i] = Bins[bin].XP_gain_mag;
+				quadZ[i] = Bins[bin].XP_gain_phase;
+				if (--i == 0)
+					break;
+			}
+		}
+		if (i == 0) {
+			polint(quadX, quadY, 1.0, &Bins[NUM_BINS - 1].XP_gain_mag);
+			polint(quadX, quadZ, 1.0, &Bins[NUM_BINS - 1].XP_gain_phase);
+		}
+		else {
+			Bins[NUM_BINS - 1].XP_gain_mag = 1.0;
+			Bins[NUM_BINS - 1].XP_gain_phase = 0.0;
+			Fault = 1;
+			if (DEBUG_PS || PsCal)
+				printf("Failure to find G0 at one\n");
+		}
+		state++;
+		break;
+	case 4:
+	case 17:
+		G0 = Bins[NUM_BINS - 1].XP_gain_mag;
+		num_bin_data = 0;
+		xp0 = -1.0;
+		xpg0 = -1.0;
+		for (bin = 0; bin < NUM_BINS; bin++) {
+			if (bin == NUM_BINS - 1) {
+				num_bin_data++;
+				Bins[NUM_BINS - 1].mag_P = 1.0;
+				Bins[NUM_BINS - 1].phase_P = - Bins[NUM_BINS - 1].XP_gain_phase;
+				Bins[NUM_BINS - 1].mag_X = 1.0;
+			}
+			else if (Bins[bin].XP_count) {
+				num_bin_data++;
+				Bins[bin].mag_P = G0 / Bins[bin].XP_gain_mag;
+				Bins[bin].phase_P = - Bins[bin].XP_gain_phase;
+				Bins[bin].mag_X = Bins[bin].XP_mag / Bins[bin].mag_P;
+				xp = Bins[bin].mag_X * Bins[bin].mag_P;
+				if (xp < xp0) {
+					if (DEBUG_PS || PsCal)
+						printf("xp is not monotonic at bin %d\n", bin);
+					Fault = 1;
+				}
+				xpg = xp * Bins[bin].XP_gain_mag;
+				if (xpg < xpg0) {
+					if (DEBUG_PS || PsCal)
+						printf("xpg is not monotonic at bin %d\n", bin);
+					Fault = 1;
+				}
+				xp0 = xp;
+				xpg0 = xpg;
+			}
+			else {
+				Bins[bin].mag_P = 0.0;
+				Bins[bin].phase_P = 0.0;
+				Bins[bin].mag_X = 0.0;
+			}
+		}
+		if (num_bin_data < NUM_BINS  * 7 / 10) {
+			if (DEBUG_PS || state == 17)
+				printf("Not enough data for corrections %d\n", num_bin_data);
+			Fault = 1;
+		}
+		if ( ! Fault) {
+#if DEBUG_PS
+			printf("Create a new predistortion\n");
+#endif
+			spline_P_size = 0;
+			for (bin = 0; bin < NUM_BINS; bin++) {
+				if (Bins[bin].XP_count) {
+					spline_P_size++;
+					spline_P_X[spline_P_size] = Bins[bin].mag_X;
+					spline_P_mag[spline_P_size] = Bins[bin].mag_P;
+					spline_P_phase[spline_P_size] = Bins[bin].phase_P;
+				}
+			}
+
+			spline(spline_P_X, spline_P_mag, spline_P_size, 2E30, 2E30, spline_P_mag2);
+			spline(spline_P_X, spline_P_phase, spline_P_size, 2E30, 2E30, spline_P_phase2);
+		}
+		state++;
+		break;
+	case 5:	
+	case 18:
+#if DEBUG_PS
+		if (Fault)
+			printf("Data for predistortion was rejected\n");
+		printf("\n");
+#endif
+		input_samples = 0;
+		state++;
+		break;
+	case 6:
+		if ( ! HOLD)
+			state = 1;
+		input_samples = 0;
+		break;
+	case 10:
+		isCalibrated = 0;
+		input_samples = 0;
+		quisk_set_key_down(1);
+		max_out_mag = 0;
+		tx_to_RX1_gain = 0;
+		starting_phase = 0;
+		ramp_counter = ramp_count0 = MILLISECS(10);
+#if DEBUG_PS
+		clip_tx = 0.0;
+#endif
+		for (bin = 0; bin < NUM_BINS; bin++) {
+			Bins[bin].table_XP = BIN_SPACING * bin;
+			Bins[bin].XP_mag = 0;
+			Bins[bin].XP_count = 0;
+			Bins[bin].XP_gain = 0;
+		}
+		tx_state = SendCW;
+		state++;
+		break;
+	case 11:
+		if (input_samples >= MILLISECS(400)) {
+			input_samples = 0;
+			state++;
+		}
+		if (input_samples >= MILLISECS(50) && cabs(amp_in_samples[0]) / CLIP32 / RX1_GAIN > 0.1) {
+			input_samples = 0;
+			state++;
+		}
+		break;
+	case 12:
+		if (input_samples >= MILLISECS(200)) {
+			state++;
+		}
+		else {
+			for (i = 0; i < nSamples; i++) {
+				in_sample = amp_in_samples[i] / CLIP32;
+				mag_in = cabs(in_sample);
+				if (tx_to_RX1_gain < mag_in)
+					tx_to_RX1_gain = mag_in;
+				out_sample = amp_out_samples[i] / CLIP32;
+				mag_out = cabs(out_sample);
+				if (max_out_mag < mag_out)
+					max_out_mag = mag_out;
+			}
+		}
+		break;
+	case 13:
+#if DEBUG_PS
+		printf("tx RX1 gain %.5f, max_out_mag %.6f\n", tx_to_RX1_gain, max_out_mag);
+#endif
+		if (fabs(tx_to_RX1_gain - RX1_GAIN) / RX1_GAIN > 1E-2)
+			printf("Measured %.6f and assumed %.6f RX1 gain differ\n", tx_to_RX1_gain, RX1_GAIN);
+		input_samples = 0;
+		state++;
+		break;
+	case 14:
+		input_samples = 0;
+		state++;
+		break;
+	case 19:
+		quisk_set_key_down(0);
+		tx_state = NothingHappens;
+		PsCal = 0;
+		if ( ! Fault)
+			isCalibrated = 1;
+		input_samples = 0;
+		state = 0;
+		break;
+	}
+}

multuspkg/__init__.py

@@ -0,0 +1 @@
+#

multuspkg/quisk_hardware.py

@@ -0,0 +1,209 @@
+# This file is part of the Multus project. See https://www.multus-sdr.com/.
+# It controls what items appear on the Hardware configuration screen and
+# provides logic for each radio.
+
+
+################ Receivers Multus CW, A transceiver from the Multus project.
+## hardware_file_name		Hardware file path, rfile
+# This is the file that contains the control logic for each radio.
+#hardware_file_name = "multuspkg/quisk_hardware.py"
+
+## widgets_file_name			Widget file path, rfile
+# This optional file adds additional controls for the radio.
+#widgets_file_name = ""
+
+## rx_max_amplitude_correct		Max ampl correct, number
+# If you get your I/Q samples from a sound card, you will need to correct the
+# amplitude and phase for inaccuracies in the analog hardware.  The correction is
+# entered using the controls from the "Rx Phase" button on the config screen.
+# You must enter a positive number.  This controls the range of the control.
+#rx_max_amplitude_correct = 0.2
+
+## rx_max_phase_correct			Max phase correct, number
+# If you get your I/Q samples from a sound card, you will need to correct the
+# amplitude and phase for inaccuracies in the analog hardware.  The correction is
+# entered using the controls from the "Rx Phase" button on the config screen.
+# You must enter a positive number.  This controls the range of the control in degrees.
+#rx_max_phase_correct = 10.0
+
+## tx_level		Tx Level, dict
+# This is the level of the Tx audio sent to SoftRock hardware after all processing as a percentage
+# number from 0 to 100.
+# The level should be below 100 to allow headroom for amplitude and phase adjustments.
+# Changes are immediate (no need to restart).
+#tx_level = {}
+
+## digital_tx_level			Digital Tx power %, integer
+# Digital modes reduce power by the percentage on the config screen.
+# This is the maximum value of the slider.
+#digital_tx_level = 100
+
+## keyer_speed		Keyer speed WPM, integer
+#  This sets the keyer speed in words per minute.
+#keyer_speed = 8
+#keyer_speed = 13
+#keyer_speed = 18
+#keyer_speed = 20
+#keyer_speed = 30
+#keyer_speed = 40
+
+## keyer_type 		Type of keyer, text choice
+# Use "Straight" mode for a straight key or a bug. Connect the tip of the key to the transceiver.
+# Otherwise, select Iambic-A or Iambic-B mode of operation.
+#keyer_type = "Straight"
+#keyer_type = "Iambic-A"
+#keyer_type = "Iambic-B"
+
+## keyer_space		Keyer space, text choice
+# This sets the type of spacing between elements or letters.
+# "Element" attempts to provide proper spacing between the elements of a character.
+# (The keyer always provides proper spacing between elements).
+# "Letter" turns on letter spacing.  This attempts to provide proper spacing between characters.
+#keyer_space = "Element"
+#keyer_space = "Letter"
+
+## keyer_weight		Keyer weight, integer choice
+# A weight of 50% means that the dit time equals the inter-element time, which is standard.
+# Reducing the weight increases the inter-element time, increasing the weight reduces it.
+#keyer_weight = 50
+#keyer_weight = 25
+#keyer_weight = 75
+
+## keyer_paddle		Keyer paddle, text choice
+# "Normal" means the left paddle is DIT. "Reverse" means the left paddle is DAH.
+#keyer_paddle = "Normal"
+#keyer_paddle = "Reverse"
+
+import configure
+import softrock
+from softrock.hardware_usb import Hardware as BaseHardware
+from softrock.hardware_usb import IN, OUT, UBYTE2, UBYTE4
+import _quisk as QS
+
+DEBUG = 0
+
+class Hardware(BaseHardware):
+  def __init__(self, app, conf):
+    # Set constant parameters here:
+    conf.usb_vendor_id = 0x16c0
+    conf.usb_product_id = 0x05dc
+    conf.si570_i2c_address = 0x55
+    conf.si570_direct_control = False
+    conf.si570_xtal_freq = 114285000
+    conf.repeater_delay = 0.25
+    self.ptt_count = 200 
+    self.ptt_on = 0
+    BaseHardware.__init__(self, app, conf)
+    QS.set_sparams(multus_cw_samples=1)
+    self.use_softrock = True
+    if DEBUG:
+      print ("The name of this radio is", configure.Settings[1])
+      print ("The hardware file type is", conf.hardware_file_type)
+  def TransferOut(self, address, message):	# message is a bytes array or an integer 0 to 255
+    if isinstance(message, int):
+      message = message.to_bytes(1, 'big')
+    if self.usb_dev:
+      self.usb_dev.ctrl_transfer(OUT, address, self.si570_i2c_address + 0x700, 0, message)
+    if DEBUG:
+      print ("USB send to 0x%X" % address, message)
+  def TransferIn(self, address, length):
+    if self.usb_dev:
+      recv = self.usb_dev.ctrl_transfer(IN, address, self.si570_i2c_address + 0x700, 0, length)
+      if DEBUG:
+        print ("USB receive from 0x%X" % address, recv)
+      return recv
+    return None
+  def open(self):
+    text = BaseHardware.open(self)
+    self.InitKeyer()
+    return text
+  def ChangeMode(self, mode):
+    ret = BaseHardware.ChangeMode(self, mode)
+    if mode in ("CWL", "CWU"):
+      cw_mode = b'C'	# Adding "b" does not affect the value sent
+      if DEBUG:
+        print ("Mode is CW")
+    else:
+      cw_mode = b'U'
+      if DEBUG:
+        print ("Mode is not CW")
+    self.TransferOut(0x70, cw_mode)
+    return ret
+  def PollCwKey(self):  # Called frequently from the sound thread to check the CW key status
+    return	# Quisk is always in Rx
+  def PollGuiControl(self):	# Called frequently from the GUI thread
+    self.ptt_count -= 1
+    if self.ptt_count <= 0:
+      self.ptt_count = 200 
+      reply = self.TransferIn(0xA5, 1)
+      if DEBUG:
+        print ("PollGuiControl got", reply)
+      if reply:
+        ptt = reply[0]	# This is 255 for error
+        if ptt in (0, 1) and ptt != self.ptt_on:
+          self.ptt_on = ptt
+          self.application.pttButton.SetValue(ptt, True)
+  def InitKeyer(self):
+    # Initialize the keyer parameters
+    conf = self.conf
+    if not hasattr(conf, "keyer_speed"):
+      conf.keyer_speed = 18
+    if not hasattr(conf, "keyer_type"):
+      conf.keyer_type = "Straight"
+    if not hasattr(conf, "keyer_space"):
+      conf.keyer_space = "Element"
+    if not hasattr(conf, "keyer_weight"):
+      conf.keyer_weight = 50
+    if not hasattr(conf, "keyer_paddle"):
+      conf.keyer_paddle = "Normal"
+    # We need to initialize in case the persistent hardware values differ from the Quisk values.
+    for name in ("keyer_speed", "keyer_type", "keyer_space", "keyer_weight", "keyer_paddle", "cwTone"):
+      self.ImmediateChange(name)
+  def ImmediateChange(self, name):
+    BaseHardware.ImmediateChange(self, name)
+    value = getattr(self.conf, name)
+    if DEBUG:
+      print ("ImmediateChange", name, value)
+    if name == "keyer_speed":
+      self.TransferOut(0x7B, value)
+    elif name == "keyer_type":
+      if value == "Straight":
+        mode = 0
+      elif value == "Iambic-A":
+        mode = 1
+      elif value == "Iambic-B":
+        mode = 2
+      else:
+        mode = 0
+      self.TransferOut(0x71, mode)
+    elif name == "keyer_space":
+      if value == "Element":
+        spacing = 0
+      elif value == "Letter":
+        spacing = 1
+      else:
+        spacing = 0
+      self.TransferOut(0x75, spacing)
+    elif name == "keyer_weight": 
+      weight = int(value)
+      self.TransferOut(0x77, weight)
+    elif name == "keyer_paddle":
+      if value == "Normal":
+        paddle = 0
+      elif value == "Reverse":
+        paddle = 1
+      else:
+        paddle = 0
+      self.TransferOut(0x73, paddle)
+    elif name == "cwTone":
+      if value < 500:
+        tone_index = 0	# 400 Hz
+      elif 500 <= value < 700:
+        tone_index = 1	# 600 Hz
+      elif 700 <= value < 900:
+        tone_index = 2	# 800 Hz
+      else:
+        tone_index = 3	# 1000 Hz
+      self.TransferOut(0x7F, tone_index)
+      
+	  

n2adr/hl2_hardware.py

@@ -1,8 +1,5 @@
 # This is the hardware control file for my shack.
-# It is for the Hermes-Lite2 5 watt output which uses only the antenna tuner.
-from __future__ import print_function
-from __future__ import absolute_import
-from __future__ import division
+# It is for the Hermes-Lite2 RF1 output and Alt Rx input.
 
 from hermes.quisk_hardware import Hardware as BaseHw
 from n2adr import station_hardware
@@ -17,7 +14,7 @@ class Hardware(BaseHw):
     self.controlbox = station_hardware.ControlBox(app, conf)	# Control my Station Control Box
     self.v2filter = station_hardware.FilterBoxV2(app, conf)	# Control V2 filter box
   def open(self):
-    if False:
+    if False:	# Start the station control GUI
       from n2adr.station_hardware import StationControlGUI
       self.GUI = StationControlGUI(self.application.main_frame, self, self.application, self.conf)
       self.GUI.Show()
@@ -29,12 +26,11 @@ class Hardware(BaseHw):
     return BaseHw.close(self)
   def ChangeFilterFrequency(self, tx_freq):
     if tx_freq and tx_freq > 0:
+      self.v2filter.SetTxFreq(tx_freq)
       if self.GUI:
         self.GUI.SetTxFreq(tx_freq)
-        self.GUI.freq_entry.ChangeValue("%.3f" % (tx_freq * 1E-6))
       else:
         self.anttuner.SetTxFreq(tx_freq)
-        self.v2filter.SetTxFreq(tx_freq)
   def ChangeFrequency(self, tx_freq, vfo_freq, source='', band='', event=None):
     self.ChangeFilterFrequency(tx_freq)
     return BaseHw.ChangeFrequency(self, tx_freq, vfo_freq, source, band, event)
@@ -42,8 +38,6 @@ class Hardware(BaseHw):
     # band is a string: "60", "40", "WWV", etc.
     ret = BaseHw.ChangeBand(self, band)
     self.anttuner.ChangeBand(band)
-    #self.lpfilter.ChangeBand(band)
-    #self.hpfilter.ChangeBand(band)
     self.v2filter.ChangeBand(band)
     self.CorrectSmeter()
     return ret

n2adr/quisk_widgets.py

@@ -1,56 +1,10 @@
-# Please do not change this widgets module for Quisk.  Instead copy
-# it to your own quisk_widgets.py and make changes there.
-#
 # This module is used to add extra widgets to the QUISK screen.
 
-from __future__ import print_function
-from __future__ import absolute_import
-from __future__ import division
-
 import wx, time
 import _quisk as QS
 
-class BottomWidgets:	# Add extra widgets to the bottom of the screen
+from hermes.quisk_widgets import BottomWidgets as BaseWidgets
+
+class BottomWidgets(BaseWidgets):	# Add extra widgets to the bottom of the screen
   def __init__(self, app, hardware, conf, frame, gbs, vertBox):
-    #self.config = conf
-    #self.hardware = hardware
-    #self.application = app
-    #start_row = app.widget_row			# The first available row
-    #start_col = app.button_start_col	# The start of the button columns
-    #b = app.QuiskCycleCheckbutton(frame, self.OnAntTuner, ('Antenna', 'Ant 0', 'Ant 1'))
-    #bw, bh = b.GetMinSize()
-    #gbs.Add(b, (start_row, start_col), (1, 2), flag=wx.EXPAND)
-    #b = app.QuiskPushbutton(frame, self.OnAntTuner, 'L+')
-    #b.Enable(0)
-    #gbs.Add(b, (start_row, start_col + 2), (1, 2), flag=wx.EXPAND)
-    #b = app.QuiskPushbutton(frame, self.OnAntTuner, 'L-')
-    #b.Enable(0)
-    #gbs.Add(b, (start_row, start_col + 4), (1, 2), flag=wx.EXPAND)
-    #b = app.QuiskPushbutton(frame, self.OnAntTuner, 'C+')
-    #b.Enable(0)
-    #gbs.Add(b, (start_row, start_col + 6), (1, 2), flag=wx.EXPAND)
-    #b = app.QuiskPushbutton(frame, self.OnAntTuner, 'C-')
-    #b.Enable(0)
-    #gbs.Add(b, (start_row, start_col + 8), (1, 2), flag=wx.EXPAND)
-    #b = app.QuiskPushbutton(frame, self.OnAntTuner, 'Save')
-    #b.Enable(0)
-    #gbs.Add(b, (start_row, start_col + 10), (1, 2), flag=wx.EXPAND)
-    #self.swr_label = app.QuiskText(frame, 'Watts 000   SWR 10.1  Zh Ind 22 Cap 33   Freq 28100 (7777)', bh)
-    #gbs.Add(self.swr_label, (start_row, start_col + 2), (1, 10), flag=wx.EXPAND)
-    #b = app.QuiskCheckbutton(frame, None, text='')
-    #gbs.Add(b, (start_row, start_col + 25), (1, 2), flag=wx.EXPAND)
-#  Example of a horizontal slider:
-#    lab = wx.StaticText(frame, -1, 'Preamp', style=wx.ALIGN_CENTER)
-#    gbs.Add(lab, (5,0), flag=wx.EXPAND)
-#    sl = wx.Slider(frame, -1, 1024, 0, 2048)	# parent, -1, initial, min, max
-#    gbs.Add(sl, (5,1), (1, 5), flag=wx.EXPAND)
-#    sl.Bind(wx.EVT_SCROLL, self.OnPreamp)
-#  def OnPreamp(self, event):
-#    print event.GetPosition()
-    self.num_rows_added = 0
-  #def OnAntTuner(self, event):
-  #  btn = event.GetEventObject()
-  #  text = btn.GetLabel()
-  #  self.hardware.OnAntTuner(text)
-  #def UpdateText(self, text):
-  #  self.swr_label.SetLabel(text)
+    BaseWidgets.__init__(self, app, hardware, conf, frame, gbs, vertBox)

n2adr/station_hardware.py

@@ -1,9 +1,5 @@
 # This file supports various hardware boxes at my shack
 
-from __future__ import print_function
-from __future__ import absolute_import
-from __future__ import division
-
 import sys, struct, math, socket, select, time, traceback, os
 import wx, wx.lib.buttons
 
@@ -411,18 +407,15 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
     self.set_HighZ = 0
     self.antnum = 0				# Antenna number 0 or 1
     self.dipole2 = AntennaControl(app, conf)	# Control the KI8BV dipole
-    if False and conf.use_rx_udp == 10:		# Hermes UDP protocol for Hermes-Lite2
-      path = 'TunerLCZ_HL2.txt'
-    else:
-      path = 'TunerLCZ.txt'
+    path = 'TunerLCZ.txt'
     if sys.platform == "win32":
-      path = 'C:/pub/' + path
+      self.path = 'C:/pub/' + path
     else:
-      path = '/home/jim/pub/' + path
-    fp = open(path, "r")
+      self.path = '/home/jim/pub/' + path
+    fp = open(self.path, "r")
     lines = fp.readlines()
     fp.close()
-    self.TunerLCZ = [(0, 0, 0, 0), (999111000, 0, 0, 0)]	# Add dummy first and last entry.
+    self.TunerLCZ = []
     for line in lines:
       freq, antL, antC, hilo = line.split()
       freq = int(freq)
@@ -451,7 +444,12 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
     pass
   def ChangeBand(self, band):
     pass
-  def FindLCZ(self, tx_freq):
+  def SaveLCZ(self):
+    fp = open(self.path, 'w')
+    for tup in self.TunerLCZ:
+      fp.write("%12d %2d %2d %d\n" % tup)
+    fp.close()
+  def FindIndexTup(self, tx_freq):
     i1 = 0
     i2 = len(self.TunerLCZ) - 1
     while 1:	# binary partition
@@ -462,20 +460,14 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
         i2 = i
       if i2 - i1 <= 1:
         break
-    # The correct setting is between i1 and i2.
+    # The correct setting is between i1 and i2, or below i1 or above i2.
     # Choose the index that is closest in frequency.
-    delta1 = tx_freq - self.TunerLCZ[i1][0]
-    delta2 = self.TunerLCZ[i2][0] - tx_freq
-    if i1 == 0:				# below the first frequency
+    if abs(tx_freq - self.TunerLCZ[i1][0]) < abs(tx_freq - self.TunerLCZ[i2][0]):
       index = i1
-    elif i2 == len(self.TunerLCZ) - 1:	# above the last frequency
+    else:
       index = i2
-    elif delta2 < delta1:	# i2 is closer
-      index = i2
-    else:			# i1 is closer
-      index = i1
-    if DEBUG: print ("AntennaTuner FindLCZ", tx_freq, i1, i2, len(self.TunerLCZ), self.TunerLCZ[index])
-    return self.TunerLCZ[index]
+    if DEBUG: print ("StatuinControlGUI FindIndexTup", tx_freq, i1, i2, len(self.TunerLCZ), index, self.TunerLCZ[index])
+    return index, self.TunerLCZ[index]
   def SetTxFreq(self, tx_freq, no_tune=False):
     #if DEBUG: print("AntennaTuner SetTxFreq", tx_freq)
     self.tx_freq = tx_freq
@@ -484,10 +476,10 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
     if tx_freq < 17000000:
       if DEBUG and self.antnum == 1: print ("antnum 0")
       self.antnum = 0
-    else:
+    elif self.dipole2:
       if DEBUG and self.antnum == 0: print ("antnum 1")
       self.antnum = 1
-    if self.antnum == 1:
+    if self.antnum == 1 and self.dipole2:
       self.dipole2.SetTxFreq(tx_freq)
     if not self.socket:
       return
@@ -496,7 +488,7 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
     if abs(self.tune_freq - tx_freq) < 5000:		# ignore small changes
       return
     self.tune_freq = tx_freq
-    f, newL, newC, newH = self.FindLCZ(tx_freq)
+    index, (f, newL, newC, newH) = self.FindIndexTup(tx_freq)
     if newH != self.set_HighZ or self.set_C != newC or self.set_L != newL:
       self.set_HighZ = newH
       self.set_C = newC
@@ -521,7 +513,8 @@ class AntennaTuner:     # Control my homebrew antenna tuner and my KI8BV dipole
         self.timer = 0
     else:
       self.timer = 0
-    self.dipole2.HeartBeat()
+    if self.dipole2:
+      self.dipole2.HeartBeat()
   def Send(self):
     try:
       self.socket.send(self.want_data)
@@ -539,26 +532,7 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     self.conf = conf
     self.data_saving = False
     self.tx_freq = 0
-    if sys.platform == "win32":
-      self.filename = 'C:/pub/TunerLCZ.tmp'
-    else:
-      self.filename = '/home/jim/pub/TunerLCZ.tmp'
-    try:
-      fp = open(self.filename, "r")
-    except FileNotFoundError:
-      lines = ()
-    else:
-      lines = fp.readlines()
-      fp.close()
-    self.TunerLCZ = [(0, 0, 0, 0), (999111000, 0, 0, 0)]	# Add dummy first and last entry.
-    for line in lines:
-      freq, antL, antC, hilo = line.split()
-      freq = int(freq)
-      antL = int(antL)
-      antC = int(antC)
-      hilo = int(hilo)
-      self.TunerLCZ.append((freq, antL, antC, hilo))
-    self.TunerLCZ.sort()
+    self.anttuner = self.hware.anttuner
     self.SetBackgroundColour('light steel blue')
     self.Bind(wx.EVT_CHAR_HOOK, self.OnKeyDown)
     sizer = wx.GridBagSizer(hgap=5, vgap=3)
@@ -568,6 +542,7 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     self.bands = []
     for band in ('160', '80', '60', '40', '30', '20', '17', '15', '12', '10'):
       b = wx.lib.buttons.GenToggleButton(self, -1, band)
+      b.Enable(False)
       b.SetUseFocusIndicator(False)
       b.SetBezelWidth(4)
       self.bands.append(b)
@@ -604,6 +579,7 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     szr = wx.BoxSizer(wx.HORIZONTAL)
     szr.AddSpacer(20)
     b = wx.lib.buttons.GenToggleButton(self, -1, "Key Down")
+    b.Enable(False)
     b.SetFont(font)
     b.SetUseFocusIndicator(False)
     b.SetBezelWidth(4)
@@ -611,6 +587,7 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     szr.Add(b)
     szr.AddSpacer(10)
     b = wx.lib.buttons.GenToggleButton(self, -1, "FilterV2 Preamp")
+    b.Enable(False)
     b.SetFont(font)
     b.SetUseFocusIndicator(False)
     b.SetBezelWidth(4)
@@ -622,16 +599,31 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     b.SetFont(font)
     szr.Add(b, flag=wx.ALIGN_CENTER_VERTICAL)
     szr.AddSpacer(10)
-    self.freq_entry = wx.TextCtrl(self, value='0000.000', style=wx.TE_PROCESS_ENTER)
+    self.freq_entry = wx.StaticText(self, -1, "0000.000")
+    #self.freq_entry = wx.TextCtrl(self, value='0000.000', style=wx.TE_PROCESS_ENTER)
     self.freq_entry.SetFont(font)
-    self.Bind(wx.EVT_TEXT_ENTER, self.OnFreqEntry, self.freq_entry)
-    szr.Add(self.freq_entry)
+    #self.Bind(wx.EVT_TEXT_ENTER, self.OnFreqEntry, self.freq_entry)
+    szr.Add(self.freq_entry, flag=wx.ALIGN_CENTER_VERTICAL)
     szr.AddSpacer(10)
-    b = wx.lib.buttons.GenButton(self, -1, "Search Freq")
+    b = wx.lib.buttons.GenButton(self, -1, "Freq-")
     b.SetUseFocusIndicator(False)
     b.SetBezelWidth(4)
     b.SetFont(font)
-    self.Bind(wx.EVT_BUTTON, self.OnButtonSearch, b)
+    self.Bind(wx.EVT_BUTTON, self.OnButtonDown, b)
+    szr.Add(b)
+    szr.AddSpacer(10)
+    b = wx.lib.buttons.GenButton(self, -1, "Freq+")
+    b.SetUseFocusIndicator(False)
+    b.SetBezelWidth(4)
+    b.SetFont(font)
+    self.Bind(wx.EVT_BUTTON, self.OnButtonUp, b)
+    szr.Add(b)
+    szr.AddSpacer(10)
+    b = wx.lib.buttons.GenButton(self, -1, "Delete")
+    b.SetUseFocusIndicator(False)
+    b.SetBezelWidth(4)
+    b.SetFont(font)
+    self.Bind(wx.EVT_BUTTON, self.OnButtonDelete, b)
     szr.Add(b)
     sizer.Add(szr, pos=(row, 0), span=(1, 3))
     if frame is None:
@@ -647,46 +639,22 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
       self.Bind(wx.EVT_TIMER, self.HeartBeat)
       self.timer.Start(100)
   def SetTxFreq(self, tx_freq):
-    #print("StationControlGUI SetTxFreq", tx_freq)
+    tx_freq = ((tx_freq + 500) // 1000) * 1000
     self.tx_freq = tx_freq
-    index, tup = self.FindIndexTup(tx_freq)
+    anttuner = self.anttuner
+    index, tup = anttuner.FindIndexTup(tx_freq)
     self.sliderL.SetValue(tup[1])
     self.sliderC.SetValue(tup[2])
     self.btnHiLo.SetValue(tup[3])
-    anttuner = self.hware.anttuner
     fff, anttuner.set_L, anttuner.set_C, anttuner.set_HighZ = tup
     if tx_freq < 17000000:
       anttuner.antnum = 0
-    else:
+    elif anttuner.dipole2:
       anttuner.antnum = 1
-    if anttuner.antnum == 1:
+    if anttuner.antnum == 1 and anttuner.dipole2:
       anttuner.dipole2.SetTxFreq(tx_freq)
     anttuner.WantData()
-  def FindIndexTup(self, tx_freq):
-    i1 = 0
-    i2 = len(self.TunerLCZ) - 1
-    while 1:	# binary partition
-      i = (i1 + i2) // 2
-      if self.TunerLCZ[i][0] < tx_freq:
-        i1 = i
-      else:
-        i2 = i
-      if i2 - i1 <= 1:
-        break
-    # The correct setting is between i1 and i2.
-    # Choose the index that is closest in frequency.
-    delta1 = tx_freq - self.TunerLCZ[i1][0]
-    delta2 = self.TunerLCZ[i2][0] - tx_freq
-    if i1 == 0:				# below the first frequency
-      index = i1
-    elif i2 == len(self.TunerLCZ) - 1:	# above the last frequency
-      index = i2
-    elif delta2 < delta1:	# i2 is closer
-      index = i2
-    else:			# i1 is closer
-      index = i1
-    if DEBUG: print ("StatuinControlGUI FindLCZ", tx_freq, i1, i2, len(self.TunerLCZ), index, self.TunerLCZ[index])
-    return index, self.TunerLCZ[index]
+    self.freq_entry.SetLabel("%.3f" % (tx_freq * 1E-6))
   def OnBtnClose(self, event):
     self.hware.anttuner.close()
     self.hware.controlbox.close()
@@ -730,16 +698,15 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     f = ((f + 500) // 1000) * 1000
     if self.hware.v2filter:
       self.hware.v2filter.SetTxFreq(f)
-    self.hware.anttuner.SetTxFreq(f, no_tune=True)
+    self.anttuner.SetTxFreq(f, no_tune=True)
     self.freq_entry.ChangeValue("%.3f" % (f * 1E-6))
     self.SetTxFreq(f)
   def OnButtonHiLo(self, event):
-    anttuner = self.hware.anttuner
     if self.btnHiLo.GetValue():
-      anttuner.set_HighZ = 1
+      self.anttuner.set_HighZ = 1
     else:
-      anttuner.set_HighZ = 0
-    anttuner.WantData()
+      self.anttuner.set_HighZ = 0
+    self.anttuner.WantData()
   def OnFreqEntry(self, event=None):
     freq = self.freq_entry.GetValue()
     freq = float(freq) * 1E6
@@ -751,11 +718,27 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     freq = int(freq + 0.1)
     self.tx_freq = freq
     self.SetTxFreq(self.tx_freq)
+  def OnButtonDown(self, event):
+    index, tup = self.anttuner.FindIndexTup(self.tx_freq)
+    index -= 1
+    if index >= 0:
+      self.SetTxFreq(self.anttuner.TunerLCZ[index][0])
+  def OnButtonUp(self, event):
+    index, tup = self.anttuner.FindIndexTup(self.tx_freq)
+    index += 1
+    if index < len(self.anttuner.TunerLCZ):
+      self.SetTxFreq(self.anttuner.TunerLCZ[index][0])
+  def OnButtonDelete(self, event):
+    index, tup = self.anttuner.FindIndexTup(self.tx_freq)
+    if self.tx_freq == self.anttuner.TunerLCZ[index][0]:
+      del self.anttuner.TunerLCZ[index]
+    if index > 0:
+      self.SetTxFreq(self.anttuner.TunerLCZ[index - 1][0])
+    elif len(self.anttuner.TunerLCZ) > 0:
+      self.SetTxFreq(self.anttuner.TunerLCZ[0][0])
+    else:
+      self.SetTxFreq(0)
   def OnButtonSave(self, event):
-    freq = self.freq_entry.GetValue()
-    freq = float(freq) * 1E6
-    freq = int(freq + 0.1)
-    self.tx_freq = freq
     L = self.sliderL.GetValue()
     C = self.sliderC.GetValue()
     if self.btnHiLo.GetValue():
@@ -763,22 +746,19 @@ class StationControlGUI(wx.Frame):    # Display a stand-alone control window for
     else:
       hilo = 0
     tup = (self.tx_freq, L, C, hilo)
-    index, t = self.FindIndexTup(self.tx_freq)
-    if self.tx_freq == self.TunerLCZ[index][0]:
-      self.TunerLCZ[index] = tup
+    index, t = self.anttuner.FindIndexTup(self.tx_freq)
+    if self.tx_freq == self.anttuner.TunerLCZ[index][0]:
+      self.anttuner.TunerLCZ[index] = tup
     else:
-      self.TunerLCZ.append(tup)
-    self.TunerLCZ.sort()
-    fp = open(self.filename, 'w')
-    for tup in self.TunerLCZ[1:-1]:	# Throw away dummy entries
-      fp.write("%12d %2d %2d %d\n" % tup)
-    fp.close()
+      self.anttuner.TunerLCZ.append(tup)
+    self.anttuner.TunerLCZ.sort()
+    self.anttuner.SaveLCZ()
   def OnSliderL(self, event=None):
-    self.hware.anttuner.set_L = self.sliderL.GetValue()
-    self.hware.anttuner.WantData()
+    self.anttuner.set_L = self.sliderL.GetValue()
+    self.anttuner.WantData()
   def OnSliderC(self, event=None):
-    self.hware.anttuner.set_C = self.sliderC.GetValue()
-    self.hware.anttuner.WantData()
+    self.anttuner.set_C = self.sliderC.GetValue()
+    self.anttuner.WantData()
   def OnBtnV2Preamp(self, event):
     if self.hware.v2filter:
       self.hware.v2filter.SetPreamp(event.GetEventObject().GetValue())
@@ -1072,21 +1052,3 @@ class AT200PC:		# Control an AT-200PC autotuner made by LDG
         self.set_C += 1
       elif text == 'C-':
         self.set_C -= 1
-
-class App(wx.App):
-  def OnInit(self):
-    if sys.path[0] != "'.'":		# Make sure the current working directory is on path
-      sys.path.insert(0, '.')
-    import quisk_conf_defaults as conf
-    import quisk_hardware_model as hardware
-    self.bottom_widgets = None
-    hardware.anttuner = AntennaTuner(self, conf)	# Control the antenna tuner
-    hardware.v2filter = FilterBoxV2(self, conf)	# Control V2 filter box
-    hardware.controlbox = ControlBox(self, conf)	# Control my Station Control Box
-    self.main_frame = frame = StationControlGUI (None, hardware, self, conf)
-    frame.Open()
-    frame.Show()
-    return True
-
-if __name__ == '__main__':
-  App().MainLoop()

perseuspkg/makefile

@@ -1,7 +1,25 @@
+MODULE_NAME := perseus
+OBJDIR := ../build/local
 
-perseus2:
-	python2 setup.py build_ext --force --inplace
+CFLAGS			:= $(shell python3 ../makefil.py --cflags)
+LDFLAGS			:= $(shell python3 ../makefil.py --ldflags)
+EXTENSION_SUFFIX	:= $(shell python3 ../makefil.py --suffix)
+TARGET := $(MODULE_NAME)$(EXTENSION_SUFFIX)
 
-perseus3:
+# Default target: Link the shared library object
+$(TARGET): perseus.o $(OBJDIR)/import_quisk_api.o
+	$(CC) -o $@ perseus.o $(LDFLAGS) $(OBJDIR)/import_quisk_api.o
+
+# Compile the C source files into object files
+perseus.o: perseus.c ../quisk.h ../filter.h
+	$(CC) -o $@ -c $< $(CFLAGS) -I..
+
+.PHONY: clean
+clean:
+	-rm -f  *.o 
+	-rm -f  $(TARGET)
+
+.PHONY: perseus-old
+perseus-old:
 	python3 setup.py build_ext --force --inplace
 

perseuspkg/perseus.c

@@ -297,11 +297,11 @@ static PyObject * open_device(PyObject * self, PyObject * args)
 
 
 	main_cleanup:
-	return PyString_FromString("ERROR");
+	return PyUnicode_FromString("ERROR");
 
 	exit_success:
 	
-	return PyString_FromString(buf128);
+	return PyUnicode_FromString(buf128);
 	
 	
 }

perseuspkg/quisk_hardware.py

@@ -18,7 +18,7 @@ from quisk_hardware_model import Hardware as BaseHardware
 DEBUG = 1
 
 # Define the name of the hardware and the items on the hardware screen (see quisk_conf_defaults.py):
-################ Receivers PerseusSDR, The PerseusSDR interface to multiple hardware SDRs.
+################ Receivers PerseusSDR, The PerseusSDR interface to multiple SDRs.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'perseuspkg/quisk_hardware.py'

pyproject.toml

@@ -0,0 +1,47 @@
+[build-system]
+requires = ["setuptools>=61.0.0"]
+build-backend = "setuptools.build_meta"
+
+[project]
+name = "quisk"
+version = "4.2.54"
+requires-python = ">=3.13"
+description = 'QUISK is a Software Defined Radio (SDR) transceiver that can control various radio hardware.'
+license = "GPL-2.0-or-later"
+license-files = ["license.txt"]
+authors = [
+  {name = "James C. Ahlstrom", email="jahlstr@gmail.com"}
+]
+keywords = ['quisk', 'sdr', 'software defined radio']
+classifiers = [
+	'Development Status :: 6 - Mature',
+	'Environment :: X11 Applications',
+	'Environment :: Win32 (MS Windows)',
+	'Intended Audience :: End Users/Desktop',
+	'Natural Language :: English',
+	'Operating System :: POSIX :: Linux',
+	'Operating System :: Microsoft :: Windows',
+	'Programming Language :: Python :: 3',
+	'Programming Language :: C',
+	'Topic :: Communications :: Ham Radio',
+]
+dynamic = ["readme"]
+
+[project.gui-scripts]
+quisk = "quisk.quisk:main"
+
+[project.urls]
+Homepage = "https://james.ahlstrom.name/quisk/index.html"
+Documentation = "https://james.ahlstrom.name/quisk/docs.html"
+Repository = "https://github.com/jimahlstrom/quisk"
+Changelog = "https://james.ahlstrom.name/quisk/CHANGELOG.txt"
+
+[tool.setuptools.package-data]
+"quisk.icons" = ["*"]
+
+[tool.setuptools]
+package-dir = {"quisk" = "."}
+packages = ["quisk", "quisk.n2adr", "quisk.softrock", "quisk.freedvpkg",
+		"quisk.hermes", "quisk.hiqsdr", "quisk.afedrinet", "quisk.soapypkg",
+		"quisk.sdrmicronpkg", "quisk.perseuspkg", "quisk.ac2yd", "quisk.multuspkg"]
+

quisk.c

@@ -3,8 +3,10 @@
 #include <stdlib.h>
 #include <math.h>
 #include <complex.h>	// Use native C99 complex type for fftw3
+#include <string.h>
 #include <fftw3.h>
 #include <sys/types.h>
+#include <stdbool.h>
 
 #ifdef MS_WINDOWS
 #include <winsock2.h>
@@ -107,12 +109,14 @@ static PyObject * pyApp;		// Application instance
 static int fft_size;			// size of fft, e.g. 1024
 int data_width;				// number of points to return as graph data; fft_size * n
 static int graph_width;			// width of the graph in pixels
-rx_mode_type rxMode;			// 0 to 13: CWL, CWU, LSB, USB, AM, FM, EXT, DGT-U, DGT-L, DGT-IQ, IMD, FDV-U, FDV-L, DGT-FM
+rx_mode_type rxMode;			// 0 to 13: CWL, CWU, LSB, USB, AM, FM, EXT, DGT_U, DGT_L, DGT_IQ, IMD, FDV_U, FDV_L, DGT_FM
 int quisk_noise_blanker;		// noise blanker level, 0 for off
 int quiskTxHoldState;			// hold Tx until the repeater frequency shift is complete
 int quisk_is_vna;			// zero for normal program, one for the VNA program
 static int py_sample_rx_bytes=2;	// number of bytes in each I or Q sample: 1, 2, 3, or 4
 static int py_sample_rx_endian;		// order of sample array: 0 == little endian; 1 == big endian
+static int py_sample_tx_bytes;		// number of bytes in each I or Q sample: 2, 3, or 4
+static int py_sample_tx_endian;		// order of sample array: 0 == little endian; 1 == big endian
 static int py_bscope_bytes;
 static int py_bscope_endian;
 static int quisk_auto_notch;	// auto notch control
@@ -150,9 +154,16 @@ static int waterfall_scroll_mode = 1;		// draw the first lines multiple times
 static int quisk_use_sidetone;		// is there a sidetone volume control?
 static int hl2_txbuf_errors;		// errors in the Hermes-Lite2 Tx buffer
 static int hl2_txbuf_state;		// state machine for errors in the Hermes-Lite2 Tx buffer
+static int softrock_correct_freq;	// gain and phase corrections for SoftRock
+static double softrock_correct_gain;
+static double softrock_correct_phase;
+static int softrock_correct_active;	// 0 No correction adjustment; 1 Manual adjustment; 2 Calculate Rx corrections
 
 static complex double PySampleBuf[SAMP_BUFFER_SIZE];	// buffer for samples returned from Python
 static int PySampleCount;				// count of samples in buffer
+static uint8_t PySampleTxBytes[SAMP_BUFFER_SIZE * 8];	// buffer for Tx sample bytes to send to the hardware
+static int PySampleTxCount;
+static bool quisk_tx_inhibit;
 
 static int multirx_data_width;			// width of graph data to return
 static int multirx_fft_width;			// size of FFT samples
@@ -179,6 +190,7 @@ int quisk_sidetoneFreq;			// frequency in hertz for the sidetone
 int quisk_start_cw_delay = 15;		// milliseconds to delay output on CW key down
 int quisk_start_ssb_delay = 100;	// milliseconds to discard output on SSB etc. key down
 static int maximum_tx_secs;		// Failsafe timeout for Tx in seconds
+static int TxRxSilenceMsec = 50;	// Play silence in msec on Tx to Rx
 static int key_is_down = 0;		// internal key state up or down
 
 static double agcReleaseGain=80;		// AGC maximum gain
@@ -211,14 +223,17 @@ static unsigned char quisk_hermes_to_pc[5 * 4];		// data received from the Herme
 static unsigned char quisk_hermeslite_response[5]; // response from Hermes-Lite commands
 unsigned int quisk_hermes_code_version = -1;		// code version returned by the Hermes hardware
 unsigned int quisk_hermes_board_id = -1;		// board ID returned by the Hermes hardware
-static double hermes_temperature;	// average temperature
-static double hermes_fwd_power;		// average forward power
-static double hermes_rev_power;		// average reverse power
-static double hermes_fwd_peak;		// peak forward power
-static double hermes_rev_peak;		// peak reverse power
-static double hermes_pa_current;	// average power amp current
-static int hermes_count_temperature;	// number of temperature samples
-static int hermes_count_current;	// number of current samples
+static double hpsdr_AIN1_avg;		// average forward power
+static double hpsdr_AIN1_peak;		// peak forward power
+static double hpsdr_AIN2_avg;		// average reverse power
+static double hpsdr_AIN2_peak;		// peak reverse power
+static double hpsdr_AIN3;		// average power amp current
+static double hpsdr_AIN4;
+static double hpsdr_AIN5;		// average temperature
+static double hpsdr_AIN6;
+static int hpsdr_AIN3_count;	// number of current samples
+static int hpsdr_AIN4_count;	// number of current samples
+static int hpsdr_AIN5_count;	// number of current samples
 static int hardware_ptt;		// hardware PTT switch
 
 int quisk_hardware_cwkey;		// hardware CW key from UDP or USB
@@ -561,15 +576,10 @@ void QuiskWavClose(struct QuiskWav * hWav)
 }
 #endif
 
-// These are used for digital voice codecs
-ty_dvoice_codec_rx  pt_quisk_freedv_rx;
-ty_dvoice_codec_tx  pt_quisk_freedv_tx;
-
-void quisk_dvoice_freedv(ty_dvoice_codec_rx rx, ty_dvoice_codec_tx tx)
+void quisk_dvoice_freedv(void)		// Not used.
 {
-	pt_quisk_freedv_rx = rx;
-	pt_quisk_freedv_tx = tx;
 }
+
 #if 0
 static int fFracDecim(double * dSamples, int nSamples, double fdecim)
 {  // fractional decimation by fdecim > 1.0
@@ -1845,7 +1855,6 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 	int i;
 	complex double cx;
 	double d, di, dd;
-	static struct AgcState Agc1 = {0.3, 16000, 0}, Agc2 = {0.3, 16000, 0};
 //static int count=0;
 //static double phase=0;
 	static struct stStorage {
@@ -1918,6 +1927,10 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 		}
 		if(bank == 0)
 			dAutoNotch(dsamples, nSamples, rit_freq, quisk_filter_srate);
+		if (ssb_squelch_enabled) {
+			ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
+			d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
+		}
 		nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
 		nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
 		nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
@@ -1937,6 +1950,10 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 		}
 		if(bank == 0)
 			dAutoNotch(dsamples, nSamples, rit_freq, quisk_filter_srate);
+		if (ssb_squelch_enabled) {
+			ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
+			d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
+		}
 		nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio12p2, 2);
 		nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand6);
 		nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
@@ -1959,7 +1976,7 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 			ssb_squelch(dsamples, nSamples, quisk_filter_srate, MeasureSquelch + bank);
 			d_delay(dsamples, nSamples, bank, SQUELCH_FFT_SIZE);
 		}
-		quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
+		//quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
 		nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p4, 2);
 		nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
 		//quisk_calc_audio_graph(pow(2, 31) - 1, NULL, dsamples, nSamples, 1);
@@ -2073,6 +2090,7 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 		MeasureSquelch[bank].squelch_active = MeasureSquelch[bank].squelch < squelch_level;
 		break;
 	case DGT_U:     // digital mode DGT-U at 48 ksps
+	case FDV_U:
 		if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) {	// filter at 6 ksps
 			quisk_filter_srate = quisk_decim_srate / 8;
 			nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
@@ -2099,6 +2117,7 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 		}
 		break;
 	case DGT_L:     // digital mode DGT-L
+	case FDV_L:
 		if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) {	// filter at 6 ksps
 			quisk_filter_srate = quisk_decim_srate / 8;
 			nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
@@ -2137,42 +2156,6 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
 			}
 		}
 		break;
-	case FDV_U:	// digital voice
-	case FDV_L:	// Extra modes added by Dave Roberts, G8KBB, June 2020.
-		quisk_check_freedv_mode();
-		// added conditional for 2400A/B mmodes
-		// current coding assumes modem sample rate is 8000 16000 or 48000
-		if( n_modem_sample_rate <= 16000 )
-			nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to16, 3);
-		if (bank == 0)
-			process_agc(&Agc1, cSamples, nSamples, 1);
-		else
-			process_agc(&Agc2, cSamples, nSamples, 1);
-		// Perhaps decimate by an additional fraction
-		if (quisk_decim_srate != 48000) {
-			dd = quisk_decim_srate / 48000.0;
-			nSamples = cFracDecim(cSamples, nSamples, dd);
-			quisk_decim_srate = 48000;
-		}
-		quisk_filter_srate = n_speech_sample_rate;
-		// added conditional for 2400A/B mmodes
-		if( n_modem_sample_rate == 8000 )
-			nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim16to8, 2);
-		// pass data decimated down to n_modem_sample_rate
-		// return data is as n_speech_sample_rate
-		if (pt_quisk_freedv_rx)
-			nSamples = (* pt_quisk_freedv_rx)(cSamples, dsamples, nSamples, bank);
-		if(bank == 0) {
-			for (i = 0; i < nSamples; i++) {
-				measure_audio_sum += dsamples[i] * dsamples[i];
-				measure_audio_count += 1;
-			}
-		}
-		// current coding assumes speech sample rate is only 8000 or 16000 - needs tweak for others
-		nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
-		if( n_speech_sample_rate == 8000 )
-			nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
-		break;
 	}
 	if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
 		measured_audio = sqrt(measure_audio_sum / measure_audio_count) / CLIP32 * 1e6;
@@ -2395,7 +2378,7 @@ int quisk_process_samples(complex double * cSamples, int nSamples)
 				quisk_sound_state.sample_rate;
 		nout = (int)dOutCounter;			// number of samples to output
 		dOutCounter -= nout;
-		playSilence = (int)(quisk_sound_state.playback_rate * 50e-3);	// Play silence after sidetone ends, number of samples
+		playSilence = (int)(quisk_sound_state.playback_rate * 1E-3 * TxRxSilenceMsec);	// Play silence after sidetone ends, number of samples
 		keyupEnvelope = 0;
 		if (quisk_active_sidetone == 2 && QUISK_CWKEY_DOWN) {	// Play sidetone instead of radio for CW
 			if (! sidetoneIsOn) {			// turn on sidetone
@@ -2467,7 +2450,8 @@ int quisk_process_samples(complex double * cSamples, int nSamples)
 			}
 	}
 
-	NoiseBlanker(cSamples, nSamples);
+	if ( ! quisk_is_key_down())
+		NoiseBlanker(cSamples, nSamples);
 
 	// Put samples into the fft input array.
 	// Thanks to WB4JFI for the code to add a third FFT buffer, July 2010.
@@ -2669,7 +2653,7 @@ int quisk_process_samples(complex double * cSamples, int nSamples)
 				multirx_cSamples[0][i] = dsamples2[i] + I * dsamples2[i];
 			process_agc(&Agc3, multirx_cSamples[0], n, 0);
 		}
-		play_sound_interface(quiskPlaybackDevices[QUISK_INDEX_SUB_RX1], n, multirx_cSamples[0], 1, 1.0);
+		play_sound_interface(quiskPlaybackDevices[QUISK_INDEX_SUB_RX1], n, multirx_cSamples[0], 1, digital_output_level);
 	}
 
 	// Perhaps decimate by an additional fraction
@@ -2679,7 +2663,7 @@ int quisk_process_samples(complex double * cSamples, int nSamples)
 		quisk_decim_srate = 48000;
 	}
 	// Process the Rx path with the WDSP library
-	nSamples = wdspFexchange0(QUISK_WDSP_RX, (double *)cSamples, nSamples);
+	nSamples = wdspFexchange0(QUISK_WDSP_RX, cSamples, nSamples);
 
 	// Interpolate the samples from 48000 sps to the play rate.
 	switch (quisk_sound_state.playback_rate / 48000) {
@@ -2707,9 +2691,6 @@ start_agc:
 	if (rxMode == EXT || rxMode == DGT_IQ) {		// Ext and DGT-IQ stereo sound
 		process_agc(&Agc1, cSamples, nSamples, 1);
 	}
-	else if (rxMode == FDV_U || rxMode == FDV_L) {	// Agc already done
-		;
-	}
 	else if (split_rxtx || multirx_play_channel >= 0) {		// separate AGC for left and right channels
 		for (i = 0; i < nSamples; i++) {
 			orig_cSamples[i] = cimag(cSamples[i]);
@@ -2867,6 +2848,8 @@ static PyObject * get_filter_rate(PyObject * self, PyObject * args)
 		break;
 	case DGT_U:     // digital modes DGT-*
 	case DGT_L:
+	case FDV_U:	 // digital voice
+	case FDV_L:
 		if (bandwidth < DGT_NARROW_FREQ)
 			filter_srate = decim_srate / 8;
 		else
@@ -2875,10 +2858,6 @@ static PyObject * get_filter_rate(PyObject * self, PyObject * args)
 	case DGT_IQ:	// digital mode at 48 ksps
 		filter_srate = decim_srate;
 		break;
-	case FDV_U:	 // digital voice
-	case FDV_L:
-		filter_srate = n_speech_sample_rate;
-		break;
 	}
 	//QuiskPrintf("Filter rate %d\n", filter_srate);
 	return PyInt_FromLong(filter_srate);
@@ -3043,6 +3022,17 @@ static PyObject * add_bscope_samples(PyObject * self, PyObject * args)
 	return Py_None;
 }
 
+static PyObject * get_tx_samples(PyObject * self, PyObject * args)
+{
+	PyObject * by;
+
+	if (!PyArg_ParseTuple (args, ""))
+		return NULL;
+	by = PyBytes_FromStringAndSize((const char *)PySampleTxBytes, PySampleTxCount);
+	PySampleTxCount = 0;
+	return by;
+}
+
 static void py_sample_start(void)
 {
 }
@@ -3065,6 +3055,83 @@ static int py_sample_read(complex double * cSamples)
 	return n;
 }
 
+static int py_sample_tx_save(complex double * cSamples, int nSamples)
+{ // These cSamples are 16-bit integers from the microphone
+	int i;
+	int16_t rr, ii;
+	uint8_t rrH, rrL, iiH, iiL;
+
+	if (PySampleTxCount + nSamples * 8 >= SAMP_BUFFER_SIZE * 8) {
+		QuiskPrintf ("py_sample_tx_save: buffer is too full\n");
+		return 0;
+	}
+	for (i = 0; i < nSamples; i++) {
+		rr = (int16_t)creal(cSamples[i]);
+		ii = (int16_t)cimag(cSamples[i]);
+		rrH = 0xFF & (rr >> 8);
+		rrL = 0xFF & rr;
+		iiH = 0xFF & (ii >> 8);
+		iiL = 0xFF & ii;
+		if (py_sample_tx_endian == 0) {		// byte order of samples is little-endian
+			switch(py_sample_tx_bytes) {
+			case 2:
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				break;
+			case 3:
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				break;
+			case 4:
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				break;
+			}
+		}
+		else {		// byte order of samples is big-endian
+			switch(py_sample_tx_bytes) {
+			case 2:
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				break;
+			case 3:
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				break;
+			case 4:
+				PySampleTxBytes[PySampleTxCount++] = rrH;
+				PySampleTxBytes[PySampleTxCount++] = rrL;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = iiH;
+				PySampleTxBytes[PySampleTxCount++] = iiL;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				PySampleTxBytes[PySampleTxCount++] = 0;
+				break;
+			}
+		}
+	}
+	return 0;
+}
+
 static PyObject * get_params(PyObject * self, PyObject * args)
 {
 	const char * name;
@@ -3084,6 +3151,10 @@ static PyObject * get_params(PyObject * self, PyObject * args)
 		return PyInt_FromLong(quisk_serial_ptt);
 	if (strcmp(name, "hl2_txbuf_errors") == 0)
 		return PyInt_FromLong(hl2_txbuf_errors);
+	if (strcmp(name, "softrock_correct_active") == 0)
+		return PyInt_FromLong((long)softrock_correct_active);
+	if (strcmp(name, "quisk_tx_inhibit") == 0)
+		return PyInt_FromLong((long)quisk_tx_inhibit);
 	Py_INCREF (Py_None);
 	return Py_None;
 }
@@ -3113,18 +3184,23 @@ static PyObject * read_fftw_wisdom(PyObject * self, PyObject * args)
 static PyObject * set_params(PyObject * self, PyObject * args, PyObject * keywds)
 {  /* Call with keyword arguments ONLY; change local parameters */
 	static char * kwlist[] = {"quisk_is_vna", "rx_bytes", "rx_endian", "read_error", "clip", 
-	"bscope_bytes", "bscope_endian", "bscope_size", "bandscopeScale", "hermes_pause", NULL} ;
+	"bscope_bytes", "bscope_endian", "bscope_size", "bandscopeScale", "hermes_pause", 
+	"tx_bytes", "tx_endian",
+	NULL} ;
 	int i, nbytes, read_error, clip, bscope_size, hermes_pause;
 
 	nbytes = read_error = clip = bscope_size = hermes_pause = -1;
-	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiiiiiiidi", kwlist,
+	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiiiiiiidiii", kwlist,
 	&quisk_is_vna, &nbytes, &py_sample_rx_endian, &read_error, &clip,
-	&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause))
+	&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause,
+	&py_sample_tx_bytes, &py_sample_tx_endian))
 		return NULL;
 	if (nbytes != -1) {
 		py_sample_rx_bytes = nbytes;
-		quisk_sample_source4(py_sample_start, py_sample_stop, py_sample_read, NULL);
+		quisk_sample_source(py_sample_start, py_sample_stop, py_sample_read);
 	}
+	if (py_sample_tx_bytes)
+		quisk_pt_sample_write = py_sample_tx_save;
 	if (read_error != -1)
 		quisk_sound_state.read_error++;
 	if (clip != -1)
@@ -3160,31 +3236,43 @@ static PyObject * get_hermes_TFRC(PyObject * self, PyObject * args)
 
 	if (!PyArg_ParseTuple (args, ""))
 		return NULL;
-	if (hermes_count_temperature > 0) {
-		hermes_temperature /= hermes_count_temperature;
-		hermes_fwd_power /= hermes_count_temperature;
+	if (hpsdr_AIN5_count > 0) {
+		hpsdr_AIN5 /= hpsdr_AIN5_count;
+		hpsdr_AIN1_avg /= hpsdr_AIN5_count;
 	}
 	else {
-		hermes_temperature  = 0.0;
-		hermes_fwd_power  = 0.0;
+		hpsdr_AIN5  = 0.0;
+		hpsdr_AIN1_avg  = 0.0;
 	}
-	if (hermes_count_current > 0) {
-		hermes_rev_power /= hermes_count_current;
-		hermes_pa_current /= hermes_count_current;
+	if (hpsdr_AIN3_count > 0) {
+		hpsdr_AIN2_avg /= hpsdr_AIN3_count;
+		hpsdr_AIN3 /= hpsdr_AIN3_count;
 	}
 	else {
-		hermes_rev_power  = 0.0;
-		hermes_pa_current  = 0.0;
+		hpsdr_AIN2_avg  = 0.0;
+		hpsdr_AIN3  = 0.0;
 	}
-	ret = Py_BuildValue("dddddd", hermes_temperature, hermes_fwd_power, hermes_rev_power, hermes_pa_current, hermes_fwd_peak, hermes_rev_peak);
-	hermes_temperature = 0;
-	hermes_fwd_power = 0;
-	hermes_rev_power = 0;
-	hermes_fwd_peak = 0;
-	hermes_rev_peak = 0;
-	hermes_pa_current = 0;
-	hermes_count_temperature = 0;
-	hermes_count_current = 0;
+	if (hpsdr_AIN4_count > 0) {
+		hpsdr_AIN4 /= hpsdr_AIN4_count;
+		hpsdr_AIN6 /= hpsdr_AIN4_count;
+	}
+	else {
+		hpsdr_AIN4  = 0.0;
+		hpsdr_AIN6  = 0.0;
+	}
+	ret = Py_BuildValue("dddddddd", hpsdr_AIN1_avg, hpsdr_AIN1_peak, hpsdr_AIN2_avg, hpsdr_AIN2_peak,
+		hpsdr_AIN3, hpsdr_AIN4, hpsdr_AIN5, hpsdr_AIN6);
+	hpsdr_AIN1_avg = 0;
+	hpsdr_AIN1_peak = 0;
+	hpsdr_AIN2_avg = 0;
+	hpsdr_AIN2_peak = 0;
+	hpsdr_AIN3 = 0;
+	hpsdr_AIN4 = 0;
+	hpsdr_AIN5 = 0;
+	hpsdr_AIN6 = 0;
+	hpsdr_AIN3_count = 0;
+	hpsdr_AIN4_count = 0;
+	hpsdr_AIN5_count = 0;
 	return ret;
 }
 
@@ -3685,6 +3773,7 @@ static int read_rx_udp10(complex double * samp)	// Read samples from UDP using t
 				//code_version = quisk_hermes_to_pc[3];
 				if ((quisk_hermes_to_pc[0] & 0x01) != 0)	// C1
 					quisk_sound_state.overrange++;
+				quisk_tx_inhibit = (quisk_hermes_to_pc[0] & 0x02) == 0;	// should limit to Hermes Lite2 only
 				hardware_ptt = buf[start] & 0x01;	// C0 bit zero is PTT
 				quisk_hardware_cwkey = (buf[start] & 0x04) >> 2;	// C0 bit two is CW key state
 				switch (hl2_txbuf_state) {
@@ -3731,19 +3820,24 @@ static int read_rx_udp10(complex double * samp)	// Read samples from UDP using t
 					quisk_set_play_state();
 				}
 			}
-			else if(dindex == 1) {	// temperature and forward power
-				hermes_temperature += quisk_hermes_to_pc[4] << 8 | quisk_hermes_to_pc[5];
+			else if(dindex == 1) {
+				hpsdr_AIN5 += quisk_hermes_to_pc[4] << 8 | quisk_hermes_to_pc[5];
 				power = quisk_hermes_to_pc[6] << 8 | quisk_hermes_to_pc[7];
-				hermes_fwd_power += power;
-				hermes_fwd_peak = fmax(hermes_fwd_peak, (double)power);
-				hermes_count_temperature++;
+				hpsdr_AIN1_avg += power;
+				hpsdr_AIN1_peak = fmax(hpsdr_AIN1_peak, (double)power);
+				hpsdr_AIN5_count++;
 			}
-			else if (dindex == 2) {	// reverse power and current
+			else if (dindex == 2) {
 				power = quisk_hermes_to_pc[8] << 8 | quisk_hermes_to_pc[9];
-				hermes_rev_power += power;
-				hermes_rev_peak = fmax(hermes_rev_peak, (double)power);
-				hermes_pa_current += quisk_hermes_to_pc[10] << 8 | quisk_hermes_to_pc[11];
-				hermes_count_current++;
+				hpsdr_AIN2_avg += power;
+				hpsdr_AIN2_peak = fmax(hpsdr_AIN2_peak, (double)power);
+				hpsdr_AIN3 += quisk_hermes_to_pc[10] << 8 | quisk_hermes_to_pc[11];
+				hpsdr_AIN3_count++;
+			}
+			else if (dindex == 3) {
+				hpsdr_AIN4 += quisk_hermes_to_pc[12] << 8 | quisk_hermes_to_pc[13];
+				hpsdr_AIN6 += quisk_hermes_to_pc[14] << 8 | quisk_hermes_to_pc[15];
+				hpsdr_AIN4_count++;
 			}
 			// convert 24-bit samples to 32-bit samples; int must be 32 bits.
 			index = start + 5;
@@ -3766,6 +3860,43 @@ static int read_rx_udp10(complex double * samp)	// Read samples from UDP using t
 			}
 		}
 	}
+	PreDistort(multirx_cSamples[0], samp, nSamples, NULL, 0);	//amp in samples, amp out samples, count
+#if 0
+	float mag_in, mag_out;
+	complex float gain;
+	complex double * amp_in_samples;
+	complex double * amp_out_samples;
+	static int up = 0;
+	static float old_mag_in = 0;
+	amp_in_samples = multirx_cSamples[0];
+	amp_out_samples = samp;
+	//if (hermes_mox_bit && quisk_multirx_count > 0) {
+	if (quisk_multirx_count == 1) {
+		for (i = 0; i < nSamples; i++) {
+			//mag_in = cabs(amp_in_samples[i]) / 0.236875 / CLIP32;
+			//mag_out = cabs(amp_out_samples[i]) / 0.088037 / CLIP32;
+			mag_in = cabs(amp_in_samples[i]) / CLIP32;
+			mag_out = cabs(amp_out_samples[i]) / CLIP32;
+			//if (0.100 - 0.01 < mag_in && mag_in < 0.100 + 0.01) {
+			//if (mag_in > .001) {
+			if (mag_in > 0.001 && mag_in < 0.234) {
+				if (up && mag_in < old_mag_in) {
+					//printf("\n");
+					up = 0;
+				}
+				else if ( ! up && mag_in > old_mag_in) {
+					//printf("\n");
+					up = 1;
+				}
+				old_mag_in = mag_in;
+				//gain = amp_out_samples[i] / amp_in_samples[i] / (0.088037 / 0.236875);
+				gain = amp_out_samples[i] / amp_in_samples[i];
+				printf("In %12.6f %8.3f  Out %12.6f %8.3f  Gain %12.6f %8.3f\n",
+				mag_in, cargf(amp_in_samples[i]), mag_out, cargf(amp_out_samples[i]), cabsf(gain), cargf(gain));
+			}
+		}
+	}
+#endif
 	if ((quisk_pc_to_hermes[3] >> 3 & 0x7) != quisk_multirx_count &&	// change in number of receivers
 		( ! quisk_multirx_count || multirx_fft_next_state == 2)) {		// wait until the current FFT is finished
 			quisk_multirx_state = 0;		// Do not change receiver count without stopping Hermes and restarting
@@ -3796,6 +3927,10 @@ static int read_rx_udp17(complex double * cSamples0)	// Read samples from UDP
 	fft_data * ptFFT;
 	fd_set fds;
 	static int block_number=0;
+	static complex double dc_average = 0;		// Average DC component in samples
+	static complex double dc_sum = 0;
+	static int dc_count = 0;
+	static time_t time0 = 0;
 	// Data from the receiver is little-endian
 
 	if ( ! rx_udp_gain_correct) {	// correct for second stage CIC decimation JIM JIM
@@ -3901,6 +4036,16 @@ static int read_rx_udp17(complex double * cSamples0)	// Read samples from UDP
 			if (xr & 0x100) {		// channel 1
 				if (quisk_invert_spectrum)		// Invert spectrum
 					sample = conj(sample);
+				// correct the DC level
+				dc_sum += sample;
+				dc_count++;
+				sample -= dc_average;
+				if (dc_count >= 48000 && time(NULL) != time0) {
+					time0 = time(NULL);
+					dc_average = dc_sum / dc_count;
+					dc_sum = dc_count = 0;
+					//printf ("New DC average %.0lf\n", cabs(dc_average));
+				}
 				// Put samples into the fft input array.
 				ptFFT = fft_data_array + fft_data_index;
 				if ( ! (xi & 0x100)) {		// zero marker for start of first block
@@ -3979,7 +4124,7 @@ static PyObject * open_rx_udp(PyObject * self, PyObject * args)
 	wVersionRequested = MAKEWORD(2, 2);
 	if (WSAStartup(wVersionRequested, &wsaData) != 0) {
 		sprintf(buf, "Failed to initialize Winsock (WSAStartup)");
-		return PyString_FromString(buf);
+		return PyUnicode_FromString(buf);
 	}
 	else {
 		cleanupWSA = 1;
@@ -4028,7 +4173,7 @@ static PyObject * open_rx_udp(PyObject * self, PyObject * args)
 	else {
 		sprintf(buf, "Failed to open socket");
 	}
-	return PyString_FromString(buf);
+	return PyUnicode_FromString(buf);
 }
 
 static PyObject * open_sound(PyObject * self, PyObject * args)
@@ -4168,6 +4313,13 @@ static PyObject * change_scan(PyObject * self, PyObject * args)	// Called from G
 	return Py_None;
 }
 
+static PyObject * softrock_corrections(PyObject * self, PyObject * args)	// Called from GUI thread
+{
+	if (!PyArg_ParseTuple (args, "i", &softrock_correct_active))
+		return NULL;
+	return Py_BuildValue("idd", softrock_correct_freq, softrock_correct_gain, softrock_correct_phase);
+}
+
 static PyObject * change_rates(PyObject * self, PyObject * args)	// Called from GUI thread
 {	// Change to new sample rates
 
@@ -4235,7 +4387,7 @@ static PyObject * mixer_set(PyObject * self, PyObject * args)
 		return NULL;
 
 	quisk_alsa_mixer_set(card_name, numid, value, err_msg, QUISK_SC_SIZE);
-	return PyString_FromString(err_msg);
+	return PyUnicode_FromString(err_msg);
 }
 
 static PyObject * pc_to_hermes(PyObject * self, PyObject * args)
@@ -4438,12 +4590,12 @@ static PyObject * ip_interfaces(PyObject * self, PyObject * args)
 				baddr |= ~netmask;
 			tup = PyTuple_New(4);
 			if (name == NULL)
-				PyTuple_SetItem(tup, 0, PyString_FromString("unnamed"));
+				PyTuple_SetItem(tup, 0, PyUnicode_FromString("unnamed"));
 			else
-				PyTuple_SetItem(tup, 0, PyString_FromString(name));
-			PyTuple_SetItem(tup, 1, PyString_FromString(Win_NtoA(ipAddr)));
-			PyTuple_SetItem(tup, 2, PyString_FromString(Win_NtoA(netmask)));
-			PyTuple_SetItem(tup, 3, PyString_FromString(Win_NtoA(baddr)));
+				PyTuple_SetItem(tup, 0, PyUnicode_FromString(name));
+			PyTuple_SetItem(tup, 1, PyUnicode_FromString(Win_NtoA(ipAddr)));
+			PyTuple_SetItem(tup, 2, PyUnicode_FromString(Win_NtoA(netmask)));
+			PyTuple_SetItem(tup, 3, PyUnicode_FromString(Win_NtoA(baddr)));
 			PyList_Append(pylist, tup);
 			Py_DECREF(tup);
 		}
@@ -4462,10 +4614,10 @@ static PyObject * ip_interfaces(PyObject * self, PyObject * args)
 		while(p) {
 			if ((p->ifa_addr) && p->ifa_addr->sa_family == AF_INET) {
 				tup = PyTuple_New(4);
-				PyTuple_SetItem(tup, 0, PyString_FromString(p->ifa_name));
-				PyTuple_SetItem(tup, 1, PyString_FromString(Lin_NtoA(p->ifa_addr)));
-				PyTuple_SetItem(tup, 2, PyString_FromString(Lin_NtoA(p->ifa_netmask)));
-				PyTuple_SetItem(tup, 3, PyString_FromString(Lin_NtoA(p->ifa_broadaddr)));
+				PyTuple_SetItem(tup, 0, PyUnicode_FromString(p->ifa_name));
+				PyTuple_SetItem(tup, 1, PyUnicode_FromString(Lin_NtoA(p->ifa_addr)));
+				PyTuple_SetItem(tup, 2, PyUnicode_FromString(Lin_NtoA(p->ifa_netmask)));
+				PyTuple_SetItem(tup, 3, PyUnicode_FromString(Lin_NtoA(p->ifa_broadaddr)));
 				PyList_Append(pylist, tup);
 				Py_DECREF(tup);
 			}
@@ -4629,6 +4781,9 @@ static PyObject * ImmediateChange(PyObject * self, PyObject * args)	// called fr
 	else if ( ! strcmp(name, "maximum_tx_secs")) {
 		maximum_tx_secs = QuiskGetConfigInt(name, 0);
 	}
+	else if ( ! strcmp(name, "TxRxSilenceMsec")) {
+		TxRxSilenceMsec = QuiskGetConfigInt(name, 50);
+	}
 	Py_INCREF (Py_None);
 	return Py_None;
 }
@@ -4952,6 +5107,135 @@ static PyObject * get_bandscope(PyObject * self, PyObject * args)	// Called by t
 	return Py_None;
 }
 
+#if 0
+static void check_channel_delay(fft_data * ptFft)
+{
+	int i, maxi, freq;
+	static int count=0, freq0;
+	static double gain=0, phase=0;
+	double d, ampl, maxa;
+	static double complex * samples1=NULL, * samples2;
+
+	if (samples1 == NULL) {
+		samples1 = (double complex *)malloc(fft_size * sizeof(double complex));
+		samples2 = (double complex *)malloc(fft_size * sizeof(double complex));
+	}
+	for (i = 0; i < fft_size; i++) {
+		samples1[i] = creal(ptFft->samples[i]);
+		samples2[i] = cimag(ptFft->samples[i]);
+	}
+	fftw_execute_dft(quisk_fft_plan, samples1, samples1);
+	fftw_execute_dft(quisk_fft_plan, samples2, samples2);
+	maxa = 0;
+	maxi = 0;
+	for (i = 0; i < fft_size / 2; i++) {
+		ampl = cabs(samples1[i]);
+		if (ampl > maxa) {
+			maxa = ampl;
+			maxi = i;
+		}
+	}
+	freq = maxi * fft_sample_rate / fft_size;
+	//freq = maxi * 44100 / fft_size;
+	gain += cabs(samples1[maxi]) / cabs(samples2[maxi]);
+	d = carg(samples1[maxi]) - carg(samples2[maxi]);
+	if (d > M_PI)
+		d -= M_PI * 2;
+	else if(d < - M_PI)
+		d += M_PI * 2;
+	phase += d;
+	count += 1;
+	if (count == 1) {
+		freq0 = freq;
+	}
+	else if (freq != freq0) {
+		gain = phase = count = 0;
+		return;
+	}
+	if (count > 10) {
+		gain /= count;
+		phase /= count;
+		//printf("Channel delay freq %6d, gain %6.3lf, phase %8.5lf radians, %6.3lf degrees\n",
+		//	freq, gain, phase, phase * 57.29578);
+		printf("%6d, %6.3lf, %6.3lf\n", freq, gain, phase * 57.29578);
+		gain = phase = count = 0;
+	}
+}
+#endif
+
+static void softrock_correct_fft(fft_data * ptFft, int fixit)
+{ // See i/Q balance in Rocky: https://www.dxatlas.com/Rocky/Advanced.asp
+	double ampl, r1, r2, i1, i2, Zr, Zi, Pwr, Gain, Phase, OutR, OutI, S1R, S1I, S2R, S2I;
+	static double tot_ga=0, tot_ph=0;
+	static int count=0;
+	double maxa = 0;
+	int i, maxi = 0, freq;
+	double complex Z, s1, s2, Hr, Hi, S1, S2;
+	double tim;
+	static double time0 = 0;
+
+	for (i = 0; i < fft_size; i++) {
+		ampl = cabs(ptFft->samples[i]);
+		if (ampl > maxa) {
+			maxa = ampl;
+			maxi = i;
+		}
+	}
+	if (maxi < fft_size / 2)
+		freq = maxi;
+	else
+		freq = maxi - fft_size;
+	freq = freq * fft_sample_rate / fft_size;
+	s1 = ptFft->samples[maxi];
+	s2 = ptFft->samples[fft_size - maxi];
+	Z = s1 * s2;
+	r1 = creal(s1);
+	r2 = creal(s2);
+	i1 = cimag(s1);
+	i2 = cimag(s2);
+	Pwr = r1 * r1 + i1 * i1 + r2 * r2 + i2 * i2;
+	Zr = creal(Z) / Pwr;
+	Zi = cimag(Z) / Pwr;
+	Gain = sqrt((1.0 + 2 * Zr) / (1.0 - 2 * Zr));
+	Phase = asin(Zi * (Gain * Gain + 1.0) / Gain);
+	tot_ga += Gain;
+	tot_ph += Phase;
+	count += 1;
+	tim = QuiskTimeSec();
+	if (tim - time0 >= 0.5) {
+		time0 = tim;
+		softrock_correct_freq = freq;
+		softrock_correct_gain = tot_ga / count;
+		softrock_correct_phase = tot_ph / count;
+		//if (softrock_correct_active)
+		//	printf("Freq domain Gain %10.7lf, Phase %10.7lf, %10.7lf, freq %d\n",
+		//		softrock_correct_gain, softrock_correct_phase, softrock_correct_phase * 57.29578, freq);
+		tot_ga = tot_ph = count = 0;
+	}
+	if (fixit) {
+		Hr = softrock_correct_gain * cos(softrock_correct_phase);
+		Hi = softrock_correct_gain * sin(softrock_correct_phase);
+		for (i = 1; i < fft_size / 2; i++) {
+			S1 = ptFft->samples[fft_size - i];
+			S2 = ptFft->samples[i];
+			S1R = creal(S1);
+			S1I = cimag(S1);
+			S2R = creal(S2);
+			S2I = cimag(S2);
+			OutR = S2R + S1R - (S2I + S1I) * Hi + (S2R - S1R) * Hr;
+			OutI = S2I - S1I + (S2I + S1I) * Hr + (S2R - S1R) * Hi;
+			ptFft->samples[i] = OutR + I * OutI;
+			S1R = creal(S2);
+			S1I = cimag(S2);
+			S2R = creal(S1);
+			S2I = cimag(S1);
+			OutR = S2R + S1R - (S2I + S1I) * Hi + (S2R - S1R) * Hr;
+			OutI = S2I - S1I + (S2I + S1I) * Hr + (S2R - S1R) * Hi;
+			ptFft->samples[fft_size - i] = OutR + I * OutI;
+		}
+	}
+}
+
 static PyObject * get_graph(PyObject * self, PyObject * args)	// Called by the GUI thread
 {
 	int i, j, k, m, n, index, ffts, ii, mm, m0, deltam;
@@ -5024,7 +5308,10 @@ static PyObject * get_graph(PyObject * self, PyObject * args)	// Called by the G
 		// Continue with FFT calculation
 		for (i = 0; i < fft_size; i++)		// multiply by window
 			ptFft->samples[i] *= fft_window[i];
+		//check_channel_delay(ptFft);
 		fftw_execute_dft(quisk_fft_plan, ptFft->samples, ptFft->samples);	// Calculate FFT
+		if (softrock_correct_active == 2)
+			softrock_correct_fft(ptFft, 0);
 		// Create RMS s-meter value at known bandwidth
 		// The pass band is (rx_tune_freq + filter_start_offset) to += bandwidth
 		// d1 is the tune frequency
@@ -5075,6 +5362,9 @@ static PyObject * get_graph(PyObject * self, PyObject * args)	// Called by the G
 			//QuiskPrintf(" %d %.4lf At %5d to %5d place %5d to %5d for block %d\n", fft_size, scan_valid, mm, m, ii, i, ptFft->block);
 		}
 		else {
+			// Zero frequency is at index fft_size / 2.
+			// There are fft_size/2 positive frequencies and fft_size/2-1 negative frequencies.
+			// The frequency at index k and (fft_size - k) are equal except for sign.
 			count_fft++;
 			k = 0;
 			for (i = fft_size / 2; i < fft_size; i++)			// Negative frequencies
@@ -5467,7 +5757,7 @@ static PyObject * Xdft(PyObject * pyseq, int inverse, int window)
 	static fftw_complex * samples;		// complex data for fft
 	static fftw_plan planF, planB;		// fft plan for fftW
 	static double * fft_window;			// window function
-	Py_complex pycx;					// Python C complex value
+	complex double pycx;
 
 	if (PySequence_Check(pyseq) != 1) {
 		PyErr_SetString (QuiskError, "DFT input data is not a sequence");
@@ -5502,22 +5792,20 @@ static PyObject * Xdft(PyObject * pyseq, int inverse, int window)
 	for (i = 0; i < size; i++) {
 		obj = PySequence_GetItem(pyseq, j);
 		if (PyComplex_Check(obj)) {
-			pycx = PyComplex_AsCComplex(obj);
+			pycx = PyComplex_RealAsDouble(obj) + I * PyComplex_ImagAsDouble(obj);
 		}
 		else if (PyFloat_Check(obj)) {
-			pycx.real = PyFloat_AsDouble(obj);
-			pycx.imag = 0;
+			pycx = PyFloat_AsDouble(obj);
 		}
 		else if (PyInt_Check(obj)) {
-			pycx.real = PyInt_AsLong(obj);
-			pycx.imag = 0;
+			pycx = PyInt_AsLong(obj);
 		}
 		else {
 			Py_XDECREF(obj);
 			PyErr_SetString (QuiskError, "DFT input data is not a complex/float/int number");
 			return NULL;
 		}
-		samples[i] = pycx.real + I * pycx.imag;
+		samples[i] = pycx;
 		if (++j >= size)
 			j = 0;
 		Py_XDECREF(obj);
@@ -5543,9 +5831,7 @@ static PyObject * Xdft(PyObject * pyseq, int inverse, int window)
 	pyseq = PyList_New(fft_size);
 	j = (size - 1) / 2;		// zero frequency in input
 	for (i = 0; i < fft_size; i++) {
-		pycx.real = creal(samples[i]);
-		pycx.imag = cimag(samples[i]);
-		PyList_SetItem(pyseq, j, PyComplex_FromCComplex(pycx));
+		PyList_SetItem(pyseq, j, PyComplex_FromDoubles(creal(samples[i]), cimag(samples[i])));
 		if (++j >= size)
 			j = 0;
 	}
@@ -5786,6 +6072,7 @@ static PyObject * record_app(PyObject * self, PyObject * args)
 	quisk_start_cw_delay = QuiskGetConfigInt("start_cw_delay", 15);
 	quisk_start_ssb_delay = QuiskGetConfigInt("start_ssb_delay", 100);
 	maximum_tx_secs = QuiskGetConfigInt("maximum_tx_secs", 0);
+	TxRxSilenceMsec = QuiskGetConfigInt("TxRxSilenceMsec", 50);
 	quisk_sound_state.sample_rate = rate;
 	fft_sample_rate = rate;
 	is_little_endian = 1;	// Test machine byte order
@@ -5899,6 +6186,8 @@ static PyMethodDef QuiskMethods[] = {
 	{"get_filter_rate", get_filter_rate, METH_VARARGS, "Return the sample rate used for the filters."},
 	{"get_tx_filter", quisk_get_tx_filter, METH_VARARGS, "Return the frequency response of the transmit filter."},
 	{"get_audio_graph", get_audio_graph, METH_VARARGS, "Return a tuple of the audio graph data."},
+	{"get_tx_samples", get_tx_samples, METH_VARARGS, "Return Tx samples to send to the hardware."},
+	{"softrock_corrections", softrock_corrections, METH_VARARGS, "Control and return SoftRock amplitude and phase corrections."},
 	{"measure_frequency", measure_frequency, METH_VARARGS, "Set the method, return the measured frequency."},
 	{"measure_audio", measure_audio, METH_VARARGS, "Set the method, return the measured audio voltage."},
 	{"get_overrange", get_overrange, METH_VARARGS, "Return the count of overrange (clip) for the ADC."},
@@ -5945,6 +6234,7 @@ static PyMethodDef QuiskMethods[] = {
 	{"XXset_transmit_mode", set_transmit_mode, METH_VARARGS, "Change the radio to transmit mode independent of key_down."},
 	{"set_volume", set_volume, METH_VARARGS, "Set the audio output volume."},
 	{"set_tx_audio", (PyCFunction)quisk_set_tx_audio, METH_VARARGS|METH_KEYWORDS, "Set the transmit audio parameters."},
+	{"get_tx_audio", (PyCFunction)quisk_get_tx_audio, METH_VARARGS, "Return some transmit audio parameters."},
 	{"is_vox", quisk_is_vox, METH_VARARGS, "return the VOX state zero or one."},
 	{"set_split_rxtx", set_split_rxtx, METH_VARARGS, "Set split for rx/tx."},
 	{"set_tune", set_tune, METH_VARARGS, "Set the tuning frequency."},
@@ -5985,13 +6275,6 @@ static PyMethodDef QuiskMethods[] = {
 	{"set_cwkey", set_hardware_cwkey, METH_VARARGS, "Change the CW key up/down state."},
 	{"set_remote_cwkey", set_remote_cwkey, METH_VARARGS, "Change the remote control CW key up/down state."},
 	{"set_PTT", set_PTT, METH_VARARGS, "Record the PTT button state."},
-	{"freedv_open", quisk_freedv_open, METH_VARARGS, "Open FreeDV."},
-	{"freedv_close", quisk_freedv_close, METH_VARARGS, "Close FreeDV."},
-	{"freedv_get_snr", quisk_freedv_get_snr, METH_VARARGS, "Return the signal to noise ratio in dB."},
-	{"freedv_get_version", quisk_freedv_get_version, METH_VARARGS, "Return the codec2 API version."},
-	{"freedv_get_rx_char", quisk_freedv_get_rx_char, METH_VARARGS, "Get text characters received from freedv."},
-	{"freedv_set_options", (PyCFunction)quisk_freedv_set_options, METH_VARARGS|METH_KEYWORDS, "Set the freedv parameters."},
-	{"freedv_set_squelch_en", quisk_freedv_set_squelch_en, METH_VARARGS, "Enable or disable FreeDV squelch."},
 	{"wdsp_set_parameter", (PyCFunction)quisk_wdsp_set_parameter, METH_VARARGS|METH_KEYWORDS, "Set parameters for the WDSP SDR library."},
 	{"tmp_record_save", tmp_record_save, METH_VARARGS, "Save the temporary recording in a WAV file."},
 	{"watfall_RgbData", watfall_RgbData, METH_VARARGS, "Return a cookie for the Waterfall pixel data."},
@@ -5999,6 +6282,8 @@ static PyMethodDef QuiskMethods[] = {
 	{"watfall_GetPixels", watfall_GetPixels, METH_VARARGS, "Write the Waterfall image to be displayed."},
 	{"write_fftw_wisdom", write_fftw_wisdom, METH_VARARGS, "Write the current fftw wisdom to the wisdom file."},
 	{"read_fftw_wisdom", read_fftw_wisdom, METH_VARARGS, "Return the current fftw wisdom as a byte array."},
+	{"tci_get_params", (PyCFunction)quisk_tci_get_params, METH_VARARGS, "Return parameters from TCI."},
+	{"tci_set_params", (PyCFunction)quisk_tci_set_params, METH_VARARGS|METH_KEYWORDS, "Set parameters for TCI."},
 // Remote Quisk control head and slave by Ben, AC2YD
 	{"start_control_head_remote_sound", quisk_start_control_head_remote_sound, METH_VARARGS, "Start running UDP remote sound on control_head."},
 	{"stop_control_head_remote_sound", quisk_stop_control_head_remote_sound, METH_VARARGS, "Stop running UDP remote sound on control_head."},

quisk.h

@@ -29,6 +29,8 @@
 
 #define QUISK_CWKEY_DOWN	(quisk_hardware_cwkey || quisk_serial_key_down || quisk_midi_cwkey || quisk_remote_cwkey)
 
+#include <stdio.h>
+#include <stdlib.h>
 #ifdef MS_WINDOWS
 #define QUISK_SHUT_RD	SD_RECEIVE
 #define QUISK_SHUT_BOTH	SD_BOTH
@@ -260,6 +262,14 @@ extern PyObject * quisk_stop_remote_radio_remote_sound(PyObject * self, PyObject
 extern int receive_graph_data(double * fft_avg);
 extern void send_graph_data(double * fft_avg, int fft_size, double zoom, double deltaf, int fft_sample_rate, double scale);
 
+// TCI server
+PyObject * quisk_tci_set_params(PyObject * self, PyObject * args, PyObject * keywds);
+PyObject * quisk_tci_get_params(PyObject * self, PyObject * args);
+void tci_send_audio(complex double * cSamples, int nSamples);
+void tci_send_iq(complex double * cSamples, int nSamples);
+int tci_get_mic(complex double * cSamples, int mic_count);
+extern uint64_t tci_tx_audio_client;	// This is ws_cli_conn_t defined in ws.h.
+
 void QuiskWavClose(struct QuiskWav *);
 int QuiskWavWriteOpen(struct QuiskWav *, char *, short, short, short, int, double);
 void QuiskWavWriteC(struct QuiskWav *, complex double *, int);
@@ -312,13 +322,11 @@ extern int quisk_active_sidetone;		// Whether and how to generate a sidetone
 extern int quisk_isFDX;				// Are we in full duplex mode?
 extern int quisk_use_serial_port;		// Are we using the serial port for CW key or PTT?
 extern play_state_t quisk_play_state;		// startup, receiving, sidetone
-extern int freedv_current_mode;			// current FreeDV mode; 700D, @)@) etc.
-extern int n_modem_sample_rate;			// Receive data, decimate to modem_sample_rate, FreeDV codec output data at speech_sample_rate
-extern int n_speech_sample_rate;		// Microphone decimate to speech_sample_rate, Freedv codec output data at modem_sample_rate, interpolate to 48000
 extern int quisk_start_cw_delay;		// milliseconds to delay output on serial or MIDI CW key down
 extern int quisk_start_ssb_delay;		// milliseconds to discard output for all modes except CW
 extern struct sound_dev * quiskPlaybackDevices[];	// array of Playback sound devices
 extern int quisk_close_file_play;
+extern double digital_output_level;
 
 extern PyObject * quisk_set_spot_level(PyObject * , PyObject *);
 extern PyObject * quisk_get_tx_filter(PyObject * , PyObject *);
@@ -337,20 +345,14 @@ extern PyObject * quisk_sound_errors(PyObject *, PyObject *);
 extern PyObject * quisk_set_file_record(PyObject *, PyObject *);
 extern PyObject * quisk_set_file_name(PyObject *, PyObject *, PyObject *);
 extern PyObject * quisk_set_tx_audio(PyObject *, PyObject *, PyObject *);
+extern PyObject * quisk_get_tx_audio(PyObject *, PyObject *);
 extern PyObject * quisk_is_vox(PyObject *, PyObject *);
 extern PyObject * quisk_set_udp_tx_correct(PyObject *, PyObject *);
 extern PyObject * quisk_set_hermes_filter(PyObject *, PyObject *);
 extern PyObject * quisk_set_alex_hpf(PyObject *, PyObject *);
 extern PyObject * quisk_set_alex_lpf(PyObject *, PyObject *);
 
-extern PyObject * quisk_freedv_open(PyObject *, PyObject *);
-extern PyObject * quisk_freedv_close(PyObject *, PyObject *);
-extern PyObject * quisk_freedv_get_snr(PyObject *, PyObject *);
-extern PyObject * quisk_freedv_get_version(PyObject *, PyObject *);
-extern PyObject * quisk_freedv_get_rx_char(PyObject *, PyObject *);
-extern PyObject * quisk_freedv_set_options(PyObject *, PyObject *, PyObject *);
 extern PyObject * quisk_set_sparams(PyObject *, PyObject *, PyObject *);
-extern PyObject * quisk_freedv_set_squelch_en(PyObject *, PyObject *);
 extern PyObject * quisk_open_key(PyObject *, PyObject *, PyObject *);
 extern PyObject * quisk_close_key(PyObject *, PyObject *);
 extern PyObject * quisk_set_sound_name(PyObject *, PyObject *);
@@ -360,7 +362,7 @@ extern PyObject * quisk_wasapi_control_midi(PyObject *, PyObject *, PyObject *);
 // WDSP interface
 #define QUISK_WDSP_RX	1
 extern PyObject * quisk_wdsp_set_parameter(PyObject *, PyObject *, PyObject *);
-extern int wdspFexchange0(int, double *, int);
+extern int wdspFexchange0(int, complex double *, int);
 
 // These function pointers are the Start/Stop/Read interface for
 // the SDR-IQ and any other C-language extension modules that return
@@ -394,7 +396,6 @@ void quisk_file_playback(complex double *, int, double);
 void quisk_tmp_playback(complex double *, int, double);
 void quisk_hermes_tx_send(int, int *);
 void quisk_udp_mic_error(char *);
-void quisk_check_freedv_mode(void);
 void quisk_calc_audio_graph(double, complex double *, double *, int, int);
 double QuiskDeltaSec(int);
 void * quisk_make_sidetone(struct sound_dev *, int);
@@ -402,12 +403,8 @@ void * quisk_make_txIQ(struct sound_dev *, int);
 int quisk_play_sidetone(struct sound_dev *);
 void quisk_set_play_state(void);
 void quisk_poll_hardware_key(void);
-
-// Functions supporting digital voice codecs
-typedef int  (* ty_dvoice_codec_rx)(complex double *, double *, int, int);
-typedef int  (* ty_dvoice_codec_tx)(complex double *, double *, int);
-extern ty_dvoice_codec_rx  pt_quisk_freedv_rx;
-extern ty_dvoice_codec_tx  pt_quisk_freedv_tx;
+void PreDistort(complex double * amp_in_samples, complex double * amp_out_samples, int nSamples, complex double * tx_samples, int num_tx);
+int CircularBuffer(int channel, complex double * cSamples, int nRead, int nWrite);
 
 // Driver function definitions=================================================
 int  quisk_read_alsa(struct sound_dev *, complex double *);
@@ -464,7 +461,7 @@ int import_quisk_api(void);	// used to initialize Quisk_API
 #define QuiskSleepMicrosec	(*(	void	(*)	(int)			)Quisk_API[5])
 #define QuiskPrintTime		(*(	void	(*)	(const char *, int)	)Quisk_API[6])
 #define quisk_sample_source	(*(	void	(*)	(ty_sample_start, ty_sample_stop, ty_sample_read)	)Quisk_API[7])
-#define quisk_dvoice_freedv	(*(	void	(*)	(ty_dvoice_codec_rx, ty_dvoice_codec_tx)        	)Quisk_API[8])
+#define quisk_dvoice_freedv	(*(	void	(*)	(void)	        	)Quisk_API[8])
 #define quisk_is_key_down	(*(	int	(*)	(void)			                                )Quisk_API[9])
 #define quisk_sample_source4	(*(	void	(*)	(ty_sample_start, ty_sample_stop, ty_sample_read, ty_sample_write)	)Quisk_API[10])
 #define strMcpy                 (*(     char *  (*)     (char *, const char *, size_t)                          )Quisk_API[11])
@@ -479,7 +476,7 @@ double	QuiskTimeSec(void);
 void	QuiskSleepMicrosec(int);
 void	QuiskPrintTime(const char *, int);
 void	quisk_sample_source(ty_sample_start, ty_sample_stop, ty_sample_read);
-void	quisk_dvoice_freedv(ty_dvoice_codec_rx, ty_dvoice_codec_tx);
+void	quisk_dvoice_freedv(void);
 int	quisk_is_key_down(void);
 void	quisk_sample_source4(ty_sample_start, ty_sample_stop, ty_sample_read, ty_sample_write);
 char *  strMcpy(char *, const char *, size_t);

quisk_conf_defaults.py

@@ -48,7 +48,7 @@ quisk_widgets = None
 
 
 
-################ Receivers SoftRock USB, Devices controlled by USB that capture samples from a sound card, and (for Tx) play samples to a sound card
+################ Receivers SoftRock USB, Sound card radios controlled by USB.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'softrock/hardware_usb.py'
@@ -133,7 +133,7 @@ digital_tx_level = 100
 
 
 
-################ Receivers SoftRock Fixed, Fixed frequency devices that capture samples from a sound card, and (for Tx) play samples to a sound card
+################ Receivers SoftRock Fixed, Fixed frequency sound card radios.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'quisk_hardware_fixed.py'
@@ -174,7 +174,7 @@ digital_tx_level = 100
 
 
 
-################ Receivers HiQSDR, The original N2ADR hardware and the improved HiQSDR using UDP
+################ Receivers HiQSDR, The original N2ADR and HiQSDR radios.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'hiqsdr/quisk_hardware.py'
@@ -258,7 +258,7 @@ sndp_active = True
 
 
 
-################ Receivers Hermes, The Hermes-Lite Project and possibly other hardware with the Hermes FPGA code.
+################ Receivers Hermes, The Hermes-Lite project by Steve Haynal.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'hermes/quisk_hardware.py'
@@ -275,7 +275,7 @@ sndp_active = True
 # You can set these options:
 
 ## use_rx_udp			Hardware type, integer choice
-# This is the type of UDP hardware.  Use 10 for the Hermes protocol.
+# This is the type of UDP hardware.  Use 10 for the Hermes Lite2 protocol.
 #use_rx_udp = 10
 
 ## udp_rx_ip		Hermes known IP, text
@@ -373,6 +373,7 @@ hermes_disable_sync = False
 # be sure to write it down. To use this address you must set "Eeprom IP Usage".
 # And you may want to set "Hermes known IP" too. Make sure the address does not conflict
 # with your DHCP server.
+# Power cycle the HL2 after changing this item.
 Hware_Hl2_EepromIP = '192.168.1.6'
 #Hware_Hl2_EepromIP = '192.168.1.241'
 
@@ -380,6 +381,7 @@ Hware_Hl2_EepromIP = '192.168.1.6'
 # This is the way the EEPROM IP address is used at power on.
 # "Ignore" means it is not used at all. "Set address" means DHCP is not used and the EEPROM IP address is used.
 # "Use DHCP first" means the EEPROM IP address is only used if DHCP fails.
+# Power cycle the HL2 after changing this item.
 Hware_Hl2_EepromIPUse = 'Ignore'
 #Hware_Hl2_EepromIPUse = 'Use DHCP first'
 #Hware_Hl2_EepromIPUse = 'Set address'
@@ -390,6 +392,7 @@ Hware_Hl2_EepromIPUse = 'Ignore'
 # at power on instead of the factory supplied MAC address 0:1c:c0:a2:13:dd. It is only used at power on,
 # and it is only used if it is marked valid by setting "Eeprom MAC usage" to "Set address".
 # If you have two HL2's, you must have a unique MAC address for each.
+# Power cycle the HL2 after changing this item.
 Hware_Hl2_EepromMAC = '0xA1 0x6B'
 #Hware_Hl2_EepromMAC = '0x4C 0x33'
 
@@ -398,11 +401,13 @@ Hware_Hl2_EepromMAC = '0xA1 0x6B'
 # the HL2 will use the last two MAC bytes from "Eeprom MAC address"; otherwise the default MAC is used.
 # If you change this item, the bit is written to the HL2.
 # The HL2 only uses these bits at power on.
+# Power cycle the HL2 after changing this item.
 Hware_Hl2_EepromMACUse = 'Ignore'
 #Hware_Hl2_EepromMACUse = 'Set address'
 
 ## hermes_TxLNA_dB			LNA during Tx dB, integer
-# During transmit the low noise Rx amplifier gain changes to this value (in dB) if the hardware supports it.
+# During transmit the low noise Rx amplifier gain changes to this value (in dB).
+# The LNA gain is -12 to 48 dB. Use -12 for Pure Signal.
 # Changes are immediate (no need to restart).
 hermes_TxLNA_dB = -12
 
@@ -419,6 +424,8 @@ hermes_PTT_hang_time = 4
 ## hermes_antenna_tuner			Antenna tuner, text choice
 # This option only applies to the Hermes Lite 2. Set this to None if you don't have a tuner.
 # Set this to "Tune" to control the Icom AH-4 compatible ATU attached to the Hermes Lite 2 end plate.
+# Note that this option only applies to the HL2 end plate, not the IO board.
+# A tuner attached to the IO board is controlled from a button on the last line of the screen.
 # Then when the Spot button is pressed with a positive power, a tune request is sent to the ATU.
 # If the Spot button is pressed with a zero power level, the tuner is set to bypass mode.
 # Changes are immediate (no need to restart).
@@ -481,6 +488,7 @@ power_meter_std_calibrations['HL2FilterE3'] = [[ 0, 0.0 ], [ 25.865384615384617,
           [ 4093.1785714285716, 7.28994845808807 ], [ 4502.496428571429, 8.820837634286566 ], [ 4952.746071428572, 10.673213537486745 ] ]
 #power_meter_std_calibrations['HL2FilterE1'] = [[0, 0.0], [9.07, 0.002], [54.98, 0.014], [148.6, 0.057],
 #          [328.0, 0.208], [611.1, 0.646], [807.0, 1.098], [982.1, 1.6], [1223.3, 2.471], [1517.7, 3.738], [1758.7, 5.02]]
+power_meter_std_calibrations['Show ADC'] = []
 
 ## Hermes_BandDict		Rx IO Bus, dict
 # The Hermes_BandDict sets the 7 bits on the J16 connector for Rx.
@@ -498,17 +506,17 @@ Hermes_BandDictEnTx = False
 #Hermes_BandDictEnTx = True
 
 
-################ Receivers Red Pitaya, The Red Pitaya Project by Pavel Demin.  This uses the Hermes FPGA code.
+################ Receivers Red Pitaya, The Red Pitaya project by Pavel Demin.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'hermes/quisk_hardware.py'
 
 ## widgets_file_name			Widget file path, rfile
 # This optional file adds additional controls for the radio.
-#widgets_file_name = ''
+#widgets_file_name = 'hermes/hpsdr_widgets.py'
 
 ## use_rx_udp			Hardware type, integer choice
-# This is the type of UDP hardware.  Use 10 for the Hermes protocol.
+# This is the type of UDP hardware.  Use 10 for the Red Pitaya using the OpenHPSDR protocol.
 #use_rx_udp = 10
 
 ## rx_udp_ip				IP change, text
@@ -576,7 +584,123 @@ Hermes_BandDictEnTx = False
 #Hermes_BandDictEnTx = True
 
 
-################ Receivers SoapySDR, The SoapySDR interface to multiple hardware SDRs.
+################ Receivers OpenHPSDR, Radios using the HPSDR Protocol 1.
+## hardware_file_name		Hardware file path, rfile
+# This is the file that contains the control logic for each radio.
+#hardware_file_name = 'hermes/quisk_hardware.py'
+
+## widgets_file_name			Widget file path, rfile
+# This optional file adds additional controls for the radio.
+#widgets_file_name = 'hermes/hpsdr_widgets.py'
+
+## use_rx_udp			Hardware type, integer choice
+# This is the type of UDP hardware.  Use 10 for hardware using the OpenHPSDR protocol 1.
+#use_rx_udp = 10
+
+## rx_udp_ip				IP change, text
+# This item should be left blank. It is used to change the IP address of the hardware to a different
+# IP once the hardware is found. Not all Hermes firmware supports changing the IP address.
+#rx_udp_ip = ""
+
+## rx_udp_port				Hardware UDP port, integer
+# This is the UDP port number of your hardware.
+#rx_udp_port = 1024
+
+## rx_udp_ip_netmask		Network netmask, text
+# This is the netmask for the network.
+#rx_udp_ip_netmask = '255.255.255.0'
+
+## tx_ip					Transmit IP, text
+# Leave this blank to use the same IP address as the receive hardware.  Otherwise, enter "disable"
+# to disable sending transmit I/Q samples, or enter the actual IP address.  You must enter "disable"
+# if you have multiple hardwares on the network, and only one should transmit. This item is normally blank.
+tx_ip = ""
+#tx_ip = "disable"
+
+## tx_audio_port			Tx audio UDP port, integer
+# This is the UDP port for transmit audio I/Q samples.  Enter zero to calculate this from the
+# base hardware port.  Otherwise enter the special custom port.
+tx_audio_port = 0
+
+## rx_udp_clock				Clock frequency Hertz, integer
+# This is the clock frequency of the hardware in Hertz.
+#rx_udp_clock = 122880000
+#rx_udp_clock = 76800000
+#rx_udp_clock = 125000000
+
+## tx_level		Tx Level, dict
+# This sets the transmit level 0 to 255 for each band.
+# The config screen has a slider 0 to 100% so you can reduce the transmit power.
+# The hardware only supports a limited adjustment range, so zero is still a small amount of power.
+tx_level = {}
+
+## digital_tx_level			Digital Tx power %, integer
+# Digital modes reduce power by the percentage on the config screen.
+# This is the maximum value of the slider.
+#digital_tx_level = 100
+
+## hermes_code_version		Hermes code version, integer
+# There can be multiple Hermes devices on a network, but Quisk can only use one of these.  If you have multiple
+# hermes devices, you can use this to specify a unique device.  Or use -1 to accept any board.
+hermes_code_version = -1
+
+## hermes_board_id			Hermes board ID, integer
+# There can be multiple Hermes devices on a network, but Quisk can only use one of these.  If you have multiple
+# hermes devices, you can use this to specify a unique device.  Or use -1 to accept any board.
+hermes_board_id = -1
+
+## Hermes_BandDict		Hermes Bus, dict
+# The Hermes_BandDict sets the 7 bits on the J16 connector.
+Hermes_BandDict = {
+	'160':0b0000001, '80':0b0000010, '60':0b0000100, '40':0b0001000, '30':0b0010000, '20':0b0100000, '15':0b1000000}
+
+## Hermes_BandDictTx		Tx IO Bus, dict
+# The Hermes_BandDictTx sets the 7 bits on the J16 connector for Tx if enabled.
+Hermes_BandDictTx = {'160':0, '80':0, '60':0, '40':0, '30':0, '20':0, '17':0, '15':0, '12':0, '10':0}
+
+## Hermes_BandDictEnTx		Enable Tx Filt, boolean
+# Enable the separate Rx and Tx settings for the J16 connector.
+Hermes_BandDictEnTx = False
+#Hermes_BandDictEnTx = True
+
+## calibrate_temp_20			PA temp ADC at 20\u2103, integer
+# This is the ADC code returned for a temperature of 20C.
+# To see the ADC code being returned, enter zero for both temperature codes.
+# Linear interpolation is used to calculate the temperature.
+# Changes are immediate (no need to restart).
+calibrate_temp_20 = 0
+
+## calibrate_temp_40			PA temp ADC at 40\u2103, integer
+# This is the ADC code returned for a temperature of 40C.
+# To see the ADC code being returned, enter zero for both temperature codes.
+# Linear interpolation is used to calculate the temperature.
+# Changes are immediate (no need to restart).
+calibrate_temp_40 = 0
+
+## calibrate_current_0			PA current ADC at zero, integer
+# This is the ADC code returned for a PA current of zero.
+# To see the ADC code being returned, enter zero for both current codes.
+# Linear interpolation is used to calculate the current.
+# Changes are immediate (no need to restart).
+calibrate_current_0 = 0
+
+## calibrate_current_1			PA current ADC at 1 amp, integer
+# This is the ADC code returned for a PA current of one amp.
+# To see the ADC code being returned, enter zero for both current codes.
+# Linear interpolation is used to calculate the current.
+# Changes are immediate (no need to restart).
+calibrate_current_1 = 0
+
+## power_meter_calib_name		Power meter calibration, text choice
+# This is the calibration table used to convert the power sensor voltage measured by the ADC to the transmit power display.
+# It is a table of ADC codes and the corresponding measured power level.  If you have a power meter, you can create your own
+# table by selecting "New". Then enter ten or more power measurements from low to full power.
+# For the Hermes Lite2 version E3 filter board, use the built-in table "HL2FilterE3".
+# Changes are immediate (no need to restart).
+power_meter_calib_name = 'HL2FilterE3'
+
+
+################ Receivers SoapySDR, The SoapySDR interface to multiple SDRs.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'soapypkg/quisk_hardware.py'
@@ -592,7 +716,7 @@ Hermes_BandDictEnTx = False
 # Further items are present in the radio dictionary with names soapy_*
 
 
-################ Receivers SdrIQ, The SDR-IQ receiver by RfSpace
+################ Receivers SdrIQ, The SDR-IQ receiver by RfSpace.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 hardware_file_name = 'quisk_hardware_sdriq.py'
@@ -623,7 +747,7 @@ sdriq_decimation = 1250
 
 
 
-################ Receivers Odyssey,   The Odyssey project using a UDP protocol similar to the HiQSDR
+################ Receivers Odyssey,   The Odyssey project.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'hiqsdr/quisk_hardware.py'
@@ -734,7 +858,7 @@ sndp_active = True
 #radio_sound_mic_boost = True
 
 
-################ Receivers Odyssey2,   The Odyssey-2 project using the HPSDR Hermes protocol
+################ Receivers Odyssey2,   The Odyssey-2 project.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'hermes/quisk_hardware.py'
@@ -749,7 +873,7 @@ sndp_active = True
 # You can set these options:
 
 ## use_rx_udp			Hardware type, integer choice
-# This is the type of UDP hardware.  Use 10 for the Hermes protocol.
+# This is the type of UDP hardware.  Use 10 for the Odyssey-2.
 #use_rx_udp = 10
 
 ## rx_udp_ip				IP change, text
@@ -818,7 +942,7 @@ Hermes_BandDictEnTx = False
 #Hermes_BandDictEnTx = True
 
 
-################ Receivers Afedri,   The Afedri SDR receiver with the Ethernet interface.
+################ Receivers Afedri,   The Afedri SDR receiver with Ethernet.
 ## hardware_file_name		Hardware file path, rfile
 # This is the file that contains the control logic for each radio.
 #hardware_file_name = 'afedrinet/quisk_hardware.py'
@@ -850,7 +974,7 @@ Hermes_BandDictEnTx = False
 #default_rf_gain = 11
 
 
-################ Receivers Control Head, Use Quisk to control a remote radio. No real hardware is used here.
+################ Receivers Control Head, Use Quisk to control a remote Quisk radio.
 ## hardware_file_name		Hardware file path, rfile
 # Quisk can be used as a control head to control a real radio located remotely.
 # This file contains the control logic for remote radios of type Hermes, SoftRock or HiQSDR.
@@ -1075,7 +1199,20 @@ digital_rx1_name = ""
 ## digital_output_level		Digital output level, number
 # This is the volume control 0.0 to 1.0 for digital playback to fldigi, etc.
 # Changes are immediate (no need to restart).
-digital_output_level = 0.7
+digital_output_level = 0.700
+
+## file_play_level		File play level, number
+# Use this to reduce the volume when playing files. For example, if the volume of the CQ
+# message is too loud, reduce the level relative to radio sound.
+# Changes are immediate (no need to restart).
+file_play_level = 1.000
+#file_play_level = 0.500
+#file_play_level = 0.100
+
+## IQ_Server_IP         Pulse server IP address, text
+# IP Address for remote PulseAudio IQ server.
+IQ_Server_IP = ""
+#IQ_Server_IP = "192.168.230.102"
 
 # Sound card names:
 #
@@ -1270,55 +1407,44 @@ dc_remove_bw = 100
 #dc_remove_bw = 200
 #dc_remove_bw = 400
 
+## fixed_tune_offset		Fixed tune offset, integer
+# If this is zero, Quisk tunes as usual. The hardware frequency (the VFO or center frequency)
+# is fixed and Quisk tunes
+# within the spectrum returned by the hardware. Otherwise enter a fixed offset between the
+# center frequency and the tuning frequency. Then tuning is done by the hardware.
+# This is useful with Softrock hardware as
+# it simplifies the corrections for amplitude and phase imbalance.
+# Changes are immediate (no need to restart).
+fixed_tune_offset = 0
+#fixed_tune_offset = 5000
+#fixed_tune_offset = 10000
+
 
 
 
 ################ Remote
-# DX cluster telent login data, thanks to DJ4CM.  Must have station_display_lines > 0.
-## dxClHost             Dx cluster host name, text
-# The Dx cluster options log into a Dx cluster server, and put station information
-# on the station window under the graph and waterfall screens.
-# dxClHost is the telnet host name.
-dxClHost = ''
-#dxClHost = 'example.host.net'
+## tci_ip		IP address for TCI, text
+# Quisk can connect to WSJT-X and other digital programs using the TCI protocol.
+# Enter the TCI server host name.
+tci_ip = "127.0.0.1"
 
-## dxClPort             Dx cluster port number, integer
-# The Dx cluster options log into a Dx cluster server, and put station information
-# on the station window under the graph and waterfall screens.
-# dxClPort is the telnet port number.
-dxClPort = 7373
-
-## user_call_sign       Call sign for Dx cluster, text
-# The Dx cluster options log into a Dx cluster server, and put station information
-# on the station window under the graph and waterfall screens.
-# user_call_sign is your call sign which may be needed for login.
-user_call_sign = ''
-
-## dxClPassword         Password for Dx cluster, text
-# The Dx cluster options log into a Dx cluster server, and put station information
-# on the station window under the graph and waterfall screens.
-# dxClPassword  is the telnet password for the server.
-dxClPassword = ''
-#dxClPassword = 'getsomedx'
-
-## dxClExpireTime       Dx cluster expire minutes, integer
-# The Dx cluster options log into a Dx cluster server, and put station information
-# on the station window under the graph and waterfall screens.
-# dxClExpireTime is the time in minutes until DX Cluster entries are removed.
-dxClExpireTime = 20
-
-## IQ_Server_IP         Pulse server IP address, text
-#IP Address for remote PulseAudio IQ server.
-IQ_Server_IP = ""
+## tci_port		IP port for TCI, integer
+# Quisk can connect to WSJT-X and other digital programs using the TCI protocol.
+# Enter the TCI port number. If the port number is zero, TCI is turned off.
+# The preferred port is 40001.
+tci_port = 0
+#tci_port = 40001
+#tci_port = 50001
 
 ## hamlib_ip            IP address for Hamlib Rig 2, text
 # You can control Quisk from Hamlib.  Set the Hamlib rig to 2 and the device for rig 2 to
 # localhost:4575.  Or choose a different name and port here.  Set the same name and port
 # in the controlling program.
 # hamlib_ip is the IP name or address.
-hamlib_ip = "localhost"
+hamlib_ip = ""
+#hamlib_ip = "localhost"
 
-## hamlib_port          IP port for Hamlib, integer
+## hamlib_port          IP port for Hamlib Rig 2, integer
 # You can control Quisk from Hamlib. For direct control, set the external program to rig 2
 # "Hamlib NET rigctl", and set the Quisk hamlib port to 4532. To use the rigctld program to control
 # Quisk, set the Quisk hamlib port to 4575. To turn off Hamlib control, set the Quisk port to zero.
@@ -1369,19 +1495,64 @@ win_hamlib_com2_name = ""
 #win_hamlib_com2_name = "COM15"
 #win_hamlib_com2_name = "COM16"
 
-## remote_radio_ip         Remote radio IP or name, text
+## remote_radio_ip         IP or name of the remote radio, text
 # Quisk can be used as a control head to control a real radio located remotely.
 # This is the IP adddress or the host name of the remote radio.
 remote_radio_ip = ""
 #remote_radio_ip = "192.168.1.56"
 
-## remote_radio_password         Password for remote radio, text
+## remote_radio_password         Password for the remote radio, text
 # Quisk can be used as a control head to control a real radio located remotely.
 # For security it requires the same password to be entered on both computers.
 # Enter a fairly long pass phrase of 20 or more characters.
 # It is only necessary to enter it once on each computer.
 remote_radio_password = ""
 
+## k4_tcp_ip			IP address for K4 TCP, text
+# This is the Quisk IP address for the TCP server implementing K4 commands.
+k4_tcp_ip = ""
+#k4_tcp_ip = "localhost"
+#k4_tcp_ip = "127.0.0.1"
+
+## k4_tcp_port			Port number for K4 TCP, integer
+# This is the Quisk port for the TCP server implementing K4 commands.
+k4_tcp_port = 0
+#k4_tcp_port = 9200
+#k4_tcp_port = 50004
+
+# DX cluster telent login data, thanks to DJ4CM.  Must have station_display_lines > 0.
+## dxClHost             Dx cluster host name, text
+# The Dx cluster options log into a Dx cluster server, and put station information
+# on the station window under the graph and waterfall screens.
+# dxClHost is the telnet host name.
+dxClHost = ''
+#dxClHost = 'example.host.net'
+
+## dxClPort             Dx cluster port number, integer
+# The Dx cluster options log into a Dx cluster server, and put station information
+# on the station window under the graph and waterfall screens.
+# dxClPort is the telnet port number.
+dxClPort = 7373
+
+## user_call_sign       Call sign for Dx cluster, text
+# The Dx cluster options log into a Dx cluster server, and put station information
+# on the station window under the graph and waterfall screens.
+# user_call_sign is your call sign which may be needed for login.
+user_call_sign = ''
+
+## dxClPassword         Password for Dx cluster, text
+# The Dx cluster options log into a Dx cluster server, and put station information
+# on the station window under the graph and waterfall screens.
+# dxClPassword  is the telnet password for the server.
+dxClPassword = ''
+#dxClPassword = 'getsomedx'
+
+## dxClExpireTime       Dx cluster expire minutes, integer
+# The Dx cluster options log into a Dx cluster server, and put station information
+# on the station window under the graph and waterfall screens.
+# dxClExpireTime is the time in minutes until DX Cluster entries are removed.
+dxClExpireTime = 20
+
 
 
 hamlib_com1_name = ""
@@ -1646,6 +1817,15 @@ start_ssb_delay = 100
 # Changes are immediate (no need to restart).
 maximum_tx_secs = 0
 
+## TxRxSilenceMsec			Tx-Rx silence msec, integer
+# When changing from transmit to receive a radio may receive either noise or the end of the transmission.
+# This may be returned as received audio, and can be heard as a loud pop.
+# This parameter plays silence for the specified number of milliseconds when changing from transmit to receive.
+# Make it long enough to zero out the pop, but not too much longer.
+# It is ignored when in full duplex mode.
+# Changes are immediate (no need to restart).
+TxRxSilenceMsec = 50
+
 ## keyupDelay			Keyup delay msecs, integer
 # This is the key hang time for semi-breakin CW. It is the time in milliseconds
 # from the last CW key release until changing to receive.
@@ -1744,23 +1924,6 @@ add_extern_demod = ""
 add_fdx_button = 0
 #add_fdx_button = 1
 
-## add_freedv_button		Add FreeDv button, integer choice
-# These buttons add up to two additional mode buttons after CW, USB, etc.
-# Set this to add the FDV mode button for digital voice:
-add_freedv_button = 1
-#add_freedv_button = 0
-
-## freedv_tx_msg			FreeDv Tx message, text
-# For freedv, this is the text message to send.
-freedv_tx_msg = ''
-#freedv_tx_msg = 'N2XXX Jim, New Jersey, USA \n'
-
-
-# This is the list of FreeDV modes and their index number.  The starting mode is the first listed.
-freedv_modes = (('Mode 1600', 0), 
-		('Mode 2400A', 3), ('Mode 2400B', 4), ('Mode 800XA', 5),
-		('Mode 700C', 6), ('Mode 700D', 7), ('Mode 2020', 8), ('Mode 700E', 13))
-
 # These are the filter bandwidths for each mode.  Quisk has built-in optimized filters
 # for these values, but you can change them if you want.
 FilterBwCW	= (200, 400, 600, 1000, 1500, 3000)
@@ -1770,7 +1933,7 @@ FilterBwFM	= (8000, 10000, 12000, 16000, 18000, 20000)
 FilterBwIMD	= FilterBwSSB
 FilterBwDGT	= (200, 400, 1500, 3200, 4800, 10000)
 FilterBwEXT	= (8000, 10000, 12000, 15000, 17000, 20000)
-FilterBwFDV	= (1500, 2000, 3000, '', '', '')
+FilterBwFDV	= (1500, 2000, 3000, '6000', '', '')
 
 # If your hardware file defines the method OnButtonPTT(self, event), then Quisk will
 # display a PTT button you can press.  The method must switch your hardware to
@@ -1890,6 +2053,8 @@ btn_text_switch = u"\u21C4"		# Character to switch left-right
 color_scheme = 'A'
 #color_scheme = 'B'
 #color_scheme = 'C'
+#color_scheme = 'DK1MI_GR'
+#color_scheme = 'DK1MI_AM'
 
 ## waterfall_palette			Waterfall colors, text choice
 # This controls the colors used in the waterfall.  The default color scheme is A, and you can change this scheme
@@ -1897,6 +2062,7 @@ color_scheme = 'A'
 waterfall_palette = 'A'
 #waterfall_palette = 'B'
 #waterfall_palette = 'C'
+#waterfall_palette = 'DK1MI_AM'
 
 
 
@@ -1931,6 +2097,8 @@ color_txline		= 'red'					# vertical line color for tx in graph
 color_rxline		= 'green'				# vertical line color for rx in graph
 color_graph_msg_fg	= 'black'				# text messages on the graph screen
 color_graph_msg_bg	= 'lemonchiffon2'		# background of text messages on the graph screen
+color_onoff		= '#77DD77'			# on/off power button
+color_onoff_text	= 'black'				# on/off power button text
 
 # This color scheme B, a dark color scheme designed by Steve Murphy, KB8RWQ.
 # Additional colors added by N2ADR.
@@ -1960,6 +2128,8 @@ color_scheme_B = {
 'color_rxline'		: 'green',
 'color_graph_msg_fg'		: 'white',
 'color_graph_msg_bg'		: '#111111',
+'color_onoff'			: '#77DD77',
+'color_onoff_text'		: 'black',
 }
 
 # This is color scheme C:
@@ -1997,12 +2167,73 @@ color_scheme_C = {
 'color_rxline'		: '#3CC918',
 'color_graph_msg_fg'	: '#000000',
 'color_graph_msg_bg'	: 'lemonchiffon2',
+'color_onoff'		: '#77DD77',
+'color_onoff_text'	: 'black',
 }
 #############################################################################################
+# This is a green dark color scheme designed by Michael
+color_scheme_DK1MI_GR = {
+    'color_bg'            : '#0F0F0F', # Lower screen background
+    'color_bg_txt'        : '#00CC00', # Lower screen text color
+    'color_graph'         : '#0F0F0F', # Graph background
+    'color_config2'       : '#008000', # color in tab row of config screen
+    'color_gl'            : '#2F4F4F', # Lines on the graph
+    'color_graphticks'    : '#CCCCCC', # Graph ticks
+    'color_graphline'     : '#00FF00', # graph data line color
+    'color_graphlabels'   : '#00FF00', # graph label color
+    'color_btn'           : '#141414', # button color
+    'color_check_btn'     : '#005500', # color of a check button when it is checked
+    'color_cycle_btn'     : '#005500', # color of a cycle button when it is checked
+    'color_adjust_btn'    : '#005500', # color of an adjustable button when it is checked
+    'color_test'          : '#005500', # color of a button used for test (turn off for tx)
+    'color_freq'          : '#208B57', # background color of frequency and s-meter
+    'color_freq_txt'      : '#00FF00', # text color of frequency display
+    'color_entry'         : '#2E2E2E', # frequency entry box
+    'color_entry_txt'     : '#00FF00', # text color of entry box
+    'color_enable'        : '#00CC00', # text color for an enabled button
+    'color_disable'       : '#FF0000', # text color for a disabled button
+    'color_popchoice'     : '#555555', # text color for button that pops up a row of buttons
+    'color_bandwidth'     : '#003300', # color for bandwidth display; thanks to WB4JFI
+    'color_txline'        : '#00CC00', # vertical line color for tx in graph
+    'color_rxline'        : '#00FF00', # vertical line color for rx in graph
+    'color_graph_msg_fg'  : '#008800', # text messages on the graph screen
+    'color_graph_msg_bg'  : '#1E1E1E', # background of text messages on the graph screen
+    'color_onoff'         : '#005500', # On/Off button
+    'color_onoff_text'    : '#FF0000', # On/Off text
+}
+#############################################################################################
+# This is an amber dark color scheme designed by Michael
+color_scheme_DK1MI_AM = {
+    'color_bg'            : '#0F0F0F', # Lower screen background
+    'color_bg_txt'        : '#FFBF00', # Lower screen text color
+    'color_graph'         : '#0F0F0F', # Graph background
+    'color_config2'       : '#FF8000', # Color in tab row of config screen
+    'color_gl'            : '#2F4F4F', # Lines on the graph
+    'color_graphticks'    : '#CCCCCC', # Graph ticks
+    'color_graphline'     : '#FFBF00', # Graph data line color
+    'color_graphlabels'   : '#FFBF00', # Graph label color
+    'color_btn'           : '#141414', # Button color
+    'color_check_btn'     : '#865E2E', # Color of a check button when it is checked
+    'color_cycle_btn'     : '#865E2E', # Color of a cycle button when it is checked
+    'color_adjust_btn'    : '#865E2E', # Color of an adjustable button when it is checked
+    'color_test'          : '#865E2E', # Color of a button used for test (turn off for tx)
+    'color_freq'          : '#FF9933', # Background color of frequency and s-meter
+    'color_freq_txt'      : '#000000', # Text color of frequency display
+    'color_entry'         : '#2E2E2E', # Frequency entry box
+    'color_entry_txt'     : '#FFBF00', # Text color of entry box
+    'color_enable'        : '#FFBF00', # Text color for an enabled button
+    'color_disable'       : '#FF0000', # Text color for a disabled button
+    'color_popchoice'     : '#555555', # Text color for button that pops up a row of buttons
+    'color_bandwidth'     : '#4B2E0F', # Color for bandwidth display
+    'color_txline'        : '#FFBF00', # Vertical line color for tx in graph
+    'color_rxline'        : '#FF8000', # Vertical line color for rx in graph
+    'color_graph_msg_fg'  : '#FF8000', # Text messages on the graph screen
+    'color_graph_msg_bg'  : '#1E1E1E', # Background of text messages on the graph screen
+    'color_onoff'         : '#865E2E', # On/Off button
+    'color_onoff_text'    : '#FF0000', # On/Off text
+}
 
-
-# These are the palettes for the waterfall.  The one used is named waterfallPallette,
-# so to use a different one, overwrite this name in your configuration file.
+# These are the palettes for the waterfall.
 waterfallPalette = (
      (  0,   0,   0,   0),
      ( 36,  85,   0, 255),
@@ -2060,6 +2291,18 @@ waterfallPaletteC = (	# from David Fainitski
 (255, 245, 99, 3)
 ) 
 
+waterfallPaletteDK1MI_AM = (	# from Michael
+    (0, 0, 0, 0),
+    (32, 25, 20, 0),
+    (64, 58, 40, 6),
+    (96, 78, 50, 16),
+    (128, 120, 85, 29),
+    (160, 144, 100, 51),
+    (192, 195, 140, 43),
+    (224, 198, 150, 35),
+    (255, 255, 191, 0)
+)
+
 # This is the data used to draw colored lines on the frequency X axis to
 # indicate CW and Phone sub-bands.  You can make it anything you want.
 # These are the colors used for sub-bands:
@@ -2085,6 +2328,26 @@ RxOnly	= '#AAAAAA'
 # Colors start at the indicated frequency and continue until the
 # next frequency.  The special color "None" turns off color.
 #
+BandPlanColors = [	# Display name and corresponding color
+  ['CW, General',	CW],
+  ['CW, Extra',		eCW],
+  ['Phone, General',	Phone],
+  ['Phone, Extra',	ePhone],
+  ['AM',		AM],
+  ['Data',		Data],
+  ['DxData',		DxData],
+  ['RTTY',		RTTY],
+  ['SSTV',		SSTV],
+  ['Packet',		Packet],
+  ['Beacons',		Beacons],
+  ['Satellite',		Satellite],
+  ['Repeater out',	Repeater],
+  ['Repeater in',	RepInput],
+  ['Simplex',		Simplex],
+  ['Rx only',		RxOnly],
+  ['Special',		Special],
+  ['Other',		Other]
+]
 
 
 

quisk_conf_kx3.py

@@ -29,9 +29,10 @@ class Hardware(BaseHardware):
     # self.hamlib_rigctld_port = 4532		# Standard rigctld control port
     # self.hamlib_poll_seconds = 0.2		# Time interval to poll for changes
   def open(self):
+    # Thanks to Peter, DC6AR, Rig ID changed from 229 to 2045.
     ret = BaseHardware.open(self)
     if not self.hamlib_connected:	# rigctld is not started.  Try to start it.
-      os.system("rigctld -m 229 -r /dev/ttyUSB0 -s 4800 & ")	# Check the baud rate menu setting
+      os.system("rigctld -m 2045 -r /dev/ttyUSB0 -s 4800 & ")	# Check the baud rate menu setting
       # If this fails, start rigctld by hand.
     return ret
 

quisk_hardware_model.py

@@ -118,15 +118,24 @@ class Hardware:
     # length is the number of samples in each block of ADC samples, and equals the FFT size.
     QS.set_params(bscope_bytes=int_size, bscope_endian=endian, bscope_size=length)
     self.application.bandscope_clock = clock
+  def InitTxSamples(self, int_size, endian, rate):	# Tx sample initialization; you must call this from your hardware __init__().
+    # int_size is the number of bytes in each I or Q sample: 2, 3, or 4
+    # endian is the order of bytes in the sample: 0 == little endian; 1 == big endian
+    # This can be called again to change the format. For example, a different number of bytes for different sample rates.
+    QS.set_tx_audio(tx_sample_rate=rate)	# Available rates are 48, 50, 96, 100, 192 ksps
+    QS.set_params(tx_bytes=int_size, tx_endian=endian)
   #def PollCwKey(self):  # Optional. Called frequently by the sound thread to check the CW key status.
   #  pass        # Do not define if not needed.
-  def StartSamples(self):	# Quisk calls this from the sound thread to start sending samples.
+  #def PollGuiControl(self):  # Optional. Called frequently by the GUI thread to change GUI settings (PTT etc.)
+  #  pass        # Do not define if not needed.
+  def StartSamples(self):	# Quisk calls this from the sound thread to start samples.
     # If you return a string, it replaces the string returned from hardware open()
     pass
-  def StopSamples(self):	# Quisk calls this from the sound thread to stop sending samples.
+  def StopSamples(self):	# Quisk calls this from the sound thread to stop samples.
     pass
   def GetRxSamples(self):	# Quisk calls this frequently from the sound thread. Poll your hardware for samples.
     # Return any available samples by calling AddRxSamples() and perhaps AddBscopeSamples() from within this method.
+    # Call GetTxSamples() from within this method to return any Tx samples that must be sent.
     pass
   def AddRxSamples(self, samples):	# Call this from within GetRxSamples() to record the Rx samples.
     # "samples" is int_size of integer I data followed by int_size of integer Q data, repeated.
@@ -140,6 +149,8 @@ class Hardware:
     # For Python 2, "samples" must be a byte array or bytes or a string.
     # The number of bytes in "samples" must equal the block length times the bytes per sample.
     QS.add_bscope_samples(samples)
+  def GetTxSamples(self):	# Call this from within GetRxSamples() to return the Tx samples to send.
+    return QS.get_tx_samples()		# Return a Python bytes object
   def GotClip(self):		# Call this to indicate that samples were received with the clip (overrange) indicator true.
     QS.set_params(clip=1)
   def GotReadError(self, print_msg, msg):	# Call this to indicate an error in receiving the Rx samples.

quisk_wdsp.c

@@ -1,61 +1,71 @@
 // This module provides C access to the WDSP SDR library.
 
 #include <Python.h>
+#include <stdio.h>
 #include <stdlib.h>
+#include <string.h>
+#include <complex.h>
 
 #define CLIP32			2147483647
 #define CLIP16			32767
 
 #define MAX_CHANNELS	32
-#define SAMPLE_BYTES	(sizeof(double) * 2)
 
 static struct _channel {
-	double * cBuf;	// save samples until nBuf reaches in_size
-	int sizeBuf;	// current size of cBuf in count of complex samples
-	int nBuf;	// current number of complex samples in cBuf
-	int in_size;	// count of complex samples needed for wdsp_fexchange0
-	int in_use;	// is this channel being used?
+	complex double * cBuf;	// circular sample buffer; save samples until nBuf reaches in_size
+	int sizeBuf;		// current size of cBuf in complex samples; a multiple of in_size
+	int nBuf;		// current number of complex samples in cBuf
+	int in_size;		// count of complex samples needed for wdsp_fexchange0
+	int in_use;		// is this channel being used?
+	int Windex;		// index of write
+	int Rindex;		// index of read
 } wdspChannel[MAX_CHANNELS];
 
 static void (*wdsp_fexchange0) (int channel, double * in, double * out, int * error);
 
-int wdspFexchange0(int channel, double * cSamples, int nSamples)
+int wdspFexchange0(int channel, complex double * cSamples, int nSamples)
 {
 // Call with an arbitrary number of input samples nSamples. cSamples is scaled to CLIP32.
-// When we have enough samples process the samples by calling wdsp. Return the samples in the same array.
-// cSamples is complex double. We treat it as interleaved Real/Imag doubles twice as long.
-// nSamples is the count of complex samples.
+// When we have enough samples process the samples by calling wdsp. Return the samples in cSamples.
+// cSamples is complex double. nSamples is the count of complex samples.
 	struct _channel * ptChannel = wdspChannel + channel;
-	int i, error;
-	double * ptD, * ptS;
+	int i, error, in_size;
 
-	if ( ! ptChannel->in_use)
+	if ( ! ptChannel->in_use) {
+		ptChannel->Windex = 0;
+		ptChannel->Rindex = 0;
+		ptChannel->nBuf = 0;
 		return nSamples;
+	}
 	if ( ! wdsp_fexchange0)
 		return nSamples;
 	if (nSamples <= 0)
 		return nSamples;
-	if (ptChannel->nBuf + nSamples >= ptChannel->sizeBuf) {
-		ptChannel->sizeBuf = ptChannel->nBuf + nSamples * 3;
-		ptChannel->cBuf = (double *)realloc(ptChannel->cBuf, ptChannel->sizeBuf * SAMPLE_BYTES);
+	in_size = ptChannel->in_size;
+	i = nSamples / in_size + 3;	// blocks need for samples plus a partial block
+	if (i * in_size > ptChannel->sizeBuf) {
+		i *= in_size;
+		ptChannel->sizeBuf = i;
+		ptChannel->cBuf = (complex double *)realloc(ptChannel->cBuf, i * sizeof(complex double));
 	}
-	ptS = cSamples;		// Copy samples from cSamples to cBuf
-	ptD = ptChannel->cBuf + ptChannel->nBuf * 2;
-	for (i = 0; i < nSamples; i++) {
-		*ptD++ = *ptS++ / CLIP32;
-		*ptD++ = *ptS++ / CLIP32;
+	for (i = 0; i < nSamples; i++) {	// Copy samples from cSamples to cBuf
+		ptChannel->cBuf[ptChannel->Windex++] = cSamples[i] / CLIP32;
+		if (ptChannel->Windex >= ptChannel->sizeBuf)
+			ptChannel->Windex = 0;
 	}
 	ptChannel->nBuf += nSamples;
 	nSamples = 0;
-	while (ptChannel->nBuf >= ptChannel->in_size) {
-		(*wdsp_fexchange0)(channel, ptChannel->cBuf, cSamples + nSamples * 2, &error);
+	while (ptChannel->nBuf >= in_size) {
+		(*wdsp_fexchange0)(channel, (double *)(ptChannel->cBuf + ptChannel->Rindex), (double *)(cSamples + nSamples), &error);
 		if (error)
 			printf("WDSP: wdsp_fexchange0 error %d\n", error);
-		nSamples += ptChannel->in_size;
-		ptChannel->nBuf -= ptChannel->in_size;
-		memmove(ptChannel->cBuf, ptChannel->cBuf + ptChannel->in_size * 2, ptChannel->nBuf * SAMPLE_BYTES);
+		ptChannel->Rindex += in_size;
+		if (ptChannel->Rindex >= ptChannel->sizeBuf)
+			ptChannel->Rindex = 0;
+		nSamples += in_size;
+		ptChannel->nBuf -= in_size;
 	}
-	for (i = 0; i < nSamples * 2; i++)
+	for (i = 0; i < nSamples; i++)
 		cSamples[i] *= CLIP32;
 	return nSamples;
 }

quisk_wdsp.py

@@ -25,16 +25,20 @@ class Cwdsp:
             pass
     else:
       try:
-        self.Lib = ctypes.CDLL("./libwdsp.so")
-        Log ("Found private wdsp")
+        self.Lib = ctypes.CDLL("./wdsp/libwdsp.so")
+        Log ("Found private wdsp in quisk/wdsp")
       except:
-        name = ctypes.util.find_library("wdsp")
-        if name:
-          try:
-            self.Lib = ctypes.CDLL(name)
-            Log ("Found public wdsp")
-          except:
-            pass
+        try:
+          self.Lib = ctypes.CDLL("./libwdsp.so")
+          Log ("Found private wdsp in quisk/")
+        except:
+          name = ctypes.util.find_library("wdsp")
+          if name:
+            try:
+              self.Lib = ctypes.CDLL(name)
+              Log ("Found public wdsp")
+            except:
+              pass
     if not self.Lib:
       Log("Wdsp was not found")
       return
@@ -73,7 +77,7 @@ class Cwdsp:
       dsp_size = 256
       QS.wdsp_set_parameter(channel, in_size=in_size)
       Lib.OpenChannel (channel, in_size, dsp_size, 48000, 48000, 48000, 0, 1,
-        ctypes.c_double(0.010), ctypes.c_double(0.025), ctypes.c_double(0.0), ctypes.c_double(0.010), 0)
+        ctypes.c_double(0.010), ctypes.c_double(0.025), ctypes.c_double(0.0), ctypes.c_double(0.010), 1)
       Lib.SetRXAShiftRun (channel, 0)
       Lib.RXANBPSetRun (channel, 0)
       Lib.SetRXAAMSQRun (channel, 0)

quisk_widgets.py

@@ -582,6 +582,7 @@ class QuiskPushbutton(QuiskButtons, wx.lib.buttons.GenButton):
     self.Bind(wx.EVT_BUTTON, self.OnButton)
     self.InitButtons(text, text_color)
     self.direction = 1
+    self.normal_brush = self.up_brush
     if use_right:
       self.Bind(wx.EVT_RIGHT_DOWN, self.OnRightDown)
       self.Bind(wx.EVT_RIGHT_UP, self.OnRightUp)
@@ -603,7 +604,12 @@ class QuiskPushbutton(QuiskButtons, wx.lib.buttons.GenButton):
       event = wx.PyEvent()
       event.SetEventObject(self)
       self.command(event)
-      
+  def SetColorNormal(self):
+    self.up_brush = self.normal_brush
+    self.Refresh()
+  def SetColor(self, color):
+    self.up_brush = wx.Brush(color)
+    self.Refresh()
 
 class QuiskRepeatbutton(QuiskButtons, wx.lib.buttons.GenButton):
   """A push button that repeats when held down."""
@@ -1549,6 +1555,8 @@ class QuiskMenu(wx.Menu):
     item.Check(is_checked)
     self.InitData(item, handler)
     return item
+  def AppendSeparator(self):
+    wx.Menu.AppendSeparator(self)
   def IsItemChecked(self, item_text):
     nid = self.item_text2id[item_text]
     menu_item = self.FindItemById(nid)

sdrtrxpkg/__init__.py

@@ -0,0 +1 @@
+#

sdrtrxpkg/quisk_hardware.py

@@ -0,0 +1,569 @@
+# SDR-TRX Quisk Configuration File
+# This file is to integrate the function of control of the SDR-TRX wint Quisk and with WSJT-X.
+# This is for control of the SDR-TRX with using the UDP port port 12346.  WSJT-X interfaces with this
+# locally using UDP on port 2237.  Capital letters begin commands to the SDR-TRX from Quisk, 
+# and lowercase letters are for commands coming from WSJT-X (through this Quisk interface).
+# This is for usinng the SDR-TRX without a 3.5 mm audio card to the soundcard of the computer,
+# and the WSJT-X software using UDP, and the data coming over UDP part 12345 too.
+# It requires the libldl/ directory from weakmon to be in the directory you run this from.
+# It also requires the WSJTXClass.py file from wsjtx_transceiver_interface to be in the lib directory
+# and the version of that class. I have supplied a pruned down versions of ft8.py and ft4.45, because we don't need much
+# other than FT8Send and FT4Send from those files.  You need to edit transceiver_config.yml to have your callsign and grid, etc.
+
+import socket
+import time
+import yaml
+import struct
+import datetime
+import sys
+import os
+from ft8 import FT8Send
+from ft4 import FT4Send
+from quisk_hardware_model import Hardware as BaseHardware
+import struct
+import collections
+import lib.WSJTXClass as WSJTXClass  # To use encoders
+
+sys.path.append(os.path.expandvars('$WEAKMON'))
+
+DEBUG_ON = True
+
+#
+# Define the name of the hardware and the items on the hardware screen (see quisk_conf_defaults.py):
+#
+
+################ Receivers SdrTrx, The SdrTrx radio by Rob Frohne, KL7NA
+## hardware_file_name		Hardware file path, rfile
+# This is the file that contains the control logic for each radio.
+#hardware_file_name = 'sdrtrxpkg/quisk_hardware.py'
+
+## widgets_file_name			Widget file path, rfile
+# This optional file adds additional controls for the radio.
+#widgets_file_name = ''
+
+## si570_xtal_freq			Si570 crystal frequency, integer
+# This is the Si570 startup frequency in Hz.  114.285MHz is the typical
+# value from the data sheet; you can use 'usbsoftrock calibrate' to find
+# the value for your device.
+si570_xtal_freq = 114285000
+
+## rx_max_amplitude_correct		Max ampl correct, number
+# If you get your I/Q samples from a sound card, you will need to correct the
+# amplitude and phase for inaccuracies in the analog hardware.  The correction is
+# entered using the controls from the "Rx Phase" button on the config screen.
+# You must enter a positive number.  This controls the range of the control.
+rx_max_amplitude_correct = 0.2
+
+## rx_max_phase_correct			Max phase correct, number
+# If you get your I/Q samples from a sound card, you will need to correct the
+# amplitude and phase for inaccuracies in the analog hardware.  The correction is
+# entered using the controls from the "Rx Phase" button on the config screen.
+# You must enter a positive number.  This controls the range of the control in degrees.
+rx_max_phase_correct = 10.0
+
+
+# SDR-TRX Hardware Control Class
+#
+
+class Hardware(BaseHardware):
+    # These are "static variable substitutes" since python doesn't have them.
+    # They are shared by all instances of the class and don't get redefinede each time a method is called.
+    # Move these into the Hardware class.
+    # Uncomment these lines if you wish to use Pulseaudio
+    name_of_sound_play = "pulse"
+
+    # Radio's Frequency limits.
+    radio_lower = 3500000
+    radio_upper = 30000000
+    PACKET_SIZE = 1444  # 4 bytes for packet number + 360 * 4 bytes for int32_t pairs
+    # Using 16 bit samples, so 2 bytes per sample.
+
+    # Set the number of Hz the signal is tuned to above the center frequency to avoid 1/f noise.
+    vfo_Center_Offset = 10000
+
+    tx_ready_wsjtx = False
+    tx_ready_wsjtx_sent = False
+    tx_now = False
+
+    # This is the open code for the WSJT server.
+    ft8_encoder = FT8Send()
+    ft4_encoder = FT4Send()
+
+    class Packet:
+        def __init__(self, data):
+            self.number = struct.unpack('<I', data[:4])[0]
+            self.pairs = [struct.unpack('<hh', data[i:i+4]) for i in range(4, len(data), 4)]
+    
+    class PacketQueue:  
+        def __init__(self):
+            self.queue = collections.deque()
+
+        def add_packet(self, packet):
+            self.queue.append(packet)
+            self.queue = collections.deque(sorted(self.queue, key=lambda p: p.number))
+
+        def check_missing_packets(self):
+            for i in range(1, len(self.queue)):
+                if self.queue[i].number != self.queue[i-1].number + 1:
+                    avg_pairs = [((self.queue[i-1].pairs[j][0] + self.queue[i].pairs[j][0]) // 2,
+                                (self.queue[i-1].pairs[j][1] + self.queue[i].pairs[j][1]) // 2)
+                                for j in range(len(self.queue[i-1].pairs))]
+                    self.queue.insert(i, Packet(struct.pack('<I', self.queue[i-1].number + 1) +
+                                                b''.join(struct.pack('<ii', pair[0], pair[1]) for pair in avg_pairs)))
+
+        def get_packets(self):
+            packets = list(self.queue)
+            self.queue.clear()
+            return packets
+
+    def open(self):
+        # Connection for WSJT-X and data
+        self.PICO_UDP_IP = None  # Will be set to the IP of the machine sending data
+        self.COMMAND_UDP_PORT = 12346  # This is the port the Pico listens on for UDP commands coming from quisk.
+        self.DATA_UDP_PORT = 12345  # This is the port the quisk listens on for UDP IQ data coming from the Pico W.
+        self.BROADCAST_PORT = 12347  # This is the port the Pico gets our IP address from.-
+        self.broadcast_sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+        self.broadcast_sock.setsockopt(socket.SOL_SOCKET, socket.SO_BROADCAST, 1)
+        self.broadcast_message = "Quisk"
+        self.broadcast_sock.setblocking(False)
+        self.isConnected = False
+        self.command_sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+        self.command_sock.setblocking(False)
+        self.data_sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+        self.data_sock.bind(('', self.DATA_UDP_PORT))
+        print("Data socket bound to IP:", self.data_sock.getsockname()[0])
+        self.data_sock.setblocking(False)
+        self.establish_connection()
+            
+        self.InitSamples(2, 0)  # Two bytes per sample, little endian.
+        self.queue = self.PacketQueue()
+        time.sleep(2)
+        # Poll for version. Should probably confirm the response on this.
+        version = str(self.get_parameter("VER"))  # The way the firmware is now, this sets it up to use the UART or UDP.
+        # UART is used if get_parameter uses the UART, and UDP is used if it uses the UDP.
+        print(version)
+        # Return an informative message for the config screen
+        t = str(version) + ". Capture from sound card %s." % self.conf.name_of_sound_capt
+        configs_file = open('transceiver_config.yml', 'r')
+        configs = yaml.load(configs_file, Loader=yaml.BaseLoader)
+
+        # Global variables
+        self.callsign = configs['callsign']
+        self.grid = configs['grid']
+        self.current_msg = ''
+        self.rx_callsign = ''
+        self.mode = 'FT8'
+        self.tx_freq = 1200
+
+        # Connection for WSJT-X
+        WSJTX_UDP_IP = "127.0.0.1"  # Assuming you are running WSJT-X on the same computer as Quisk...
+        WSJTX_UDP_PORT = 2237
+        self.wsjtx_sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+        self.wsjtx_sock.bind((WSJTX_UDP_IP, WSJTX_UDP_PORT))
+        self.wsjtx_sock.setblocking(False)
+
+        self.StartSamples()
+        return t
+
+    def close(self):
+        # Called once to close the Hardware
+        self.get_parameter("RX") # Make sure we are in RX mode.
+        time.sleep(1)  # Wait for RX to be set.
+        self.StopSamples()
+        self.command_sock.close()
+        self.wsjtx_sock.close()
+        self.broadcast_sock.close()
+
+    def ChangeFrequency(self, tune, vfo, source='', band='', event=None):
+        # Called whenever quisk requests a frequency change.
+        # This sends the FREQ command  dto set the centre frequency of the radio,
+        # and will also move the 'tune' frequency (the section within the RX passband
+        # which is to be demodulated) if it falls outside the passband (+/- sample_rate/2).
+        print("Setting VFO to %d, with tune to %d." % (vfo, tune))
+        print("Setting VFO to %d." % vfo)
+        if vfo < self.radio_lower:
+            vfo = self.radio_lower
+            print("Outside range! Setting to %d" % self.radio_lower)
+
+        if vfo > self.radio_upper:
+            vfo = self.radio_upper
+            print("Outside range! Setting to %d" % self.radio_upper)
+
+        # print("sample_rate =", sample_rate)  # I believe sample_rate comes from quisk.
+        # If the tune frequency is outside the RX bandwidth, set it to somewhere within that bandwidth.
+        if tune > (vfo + sample_rate / 2) or tune < (vfo - sample_rate / 2):
+            tune = vfo + self.vfo_Center_Offset
+            print("Bringing tune frequency back into the RX bandwidth.")
+
+        # success = self.set_parameter("FREQ",str(vfo))
+        self.set_parameter("FREQ", str(vfo))
+        self.set_parameter("TX_FREQ", str(tune))
+
+        return tune, vfo
+
+    def OnButtonPTT(self, event):
+        btn = event.GetEventObject()
+        if btn.GetValue():
+            self.get_parameter("TX")
+            print("TX")
+        else:
+            self.get_parameter("RX")
+            print("RX")
+        return BaseHardware.OnButtonPTT(self, event)
+
+
+
+    def establish_connection(self):
+        while not self.isConnected:
+            print("isConnected is: ", self.isConnected)
+            # Check if a connection is established
+            # Wait for data to arrive and set PICO_UDP_IP to the sender's IP
+            while self.PICO_UDP_IP is None:  #  This needs to be fixed.  It continues to send broadcast packets.
+                try:
+                    print("Sending broadcast message...")
+                    self.broadcast_sock.sendto(self.broadcast_message.encode(), ('<broadcast>', self.BROADCAST_PORT))
+                    time.sleep(1)  # Wait for 1 second before sending the next broadcast message
+                except Exception as e:
+                    print("Broadcast message did not work: {0}".format(e))
+                    break
+                try:
+                    data, addr = self.data_sock.recvfrom(self.PACKET_SIZE)  # Adjust the buffer size as needed
+                    self.PICO_UDP_IP = addr[0]
+                    print("PICO_UDP_IP set to:", self.PICO_UDP_IP)
+                    # if data == self.broadcast_message:
+                    self.isConnected = True
+                except socket.error:
+                    if self.PICO_UDP_IP is not None:
+                        print("Connection established with Pico W at IP:", self.PICO_UDP_IP, "but exception occurred.")
+                        break  # Exit the loop if self.PICO_UDP_IP is not None
+                    pass  # No data available yet
+
+    def GetRxSamples(self):
+        # Initialize packet counter and time marker
+        # This routine is made for FT8 and FT4, where we need to send a packet every 15 seconds.
+        # The Pico W sending the data has a sample rate of 48046 samples per second > 48 ks/s.
+        # We made the packets so they carry 180 samples per packet.  That gives 4000 packets 
+        # per 15 seconds.  We discard any packets after that four times a minute at 59, 14, 29, 
+        # and 44 seconds.  This should make four ticks per minute, and we should be able to ignore
+        # # them if we are doing FT8 or FT4.  Something else might be more optimum for other modes.
+
+        packet_counter = 0
+        next_time_marker = 0
+        send_packets = False
+
+        while True:
+            # Check if we've reached the next time marker
+            current_time = time.time()
+            current_seconds = current_time % 60
+
+            # Define the time markers
+            if 59 <= current_seconds < 60:
+                next_time_marker = current_time + 1 - current_seconds
+                send_packets = True
+            elif 14 <= current_seconds < 15:
+                next_time_marker = current_time + 1 - (current_seconds - 14)
+                send_packets = True
+            elif 29 <= current_seconds < 30:
+                next_time_marker = current_time + 1 - (current_seconds - 29)
+                send_packets = True
+            elif 44 <= current_seconds < 45:
+                next_time_marker = current_time + 1 - (current_seconds - 44)
+                send_packets = True
+
+            # If we've reached the next time marker, reset the packet counter
+            if current_time >= next_time_marker and send_packets:
+                packet_counter = 0
+                send_packets = False
+
+            try:
+                data, addr = self.data_sock.recvfrom(self.PACKET_SIZE)
+                self.no_data_repeat = 0
+            except socket.error:
+                # Send a broadcast message to the Pico W to get the IP address in case it has reset.
+                self.broadcast_sock.sendto(self.broadcast_message.encode(), ('<broadcast>', self.BROADCAST_PORT))
+                break  # No more packets available
+
+            packet = self.Packet(data)
+            self.queue.add_packet(packet)
+
+            # Check for missing packets and fill them in
+            self.queue.check_missing_packets()
+
+            # Get the packets and pass them to the processing thread
+            packets = self.queue.get_packets()
+
+            for packet in packets:
+                data = b''.join(struct.pack('<hh', pair[0], pair[1]) for pair in packet.pairs)
+                # if len(data) % 4 != 0:  # Each I/Q pair is now 4 bytes (2 * 2 bytes)
+                #     self.GotReadError(self, DEBUG_ON, "Error: Mismatched number of I and Q samples")
+                #     continue
+                            # If we've already sent 4000 packets since the last time marker, discard the rest
+                if packet_counter >= 4000:
+                    continue
+
+                self.AddRxSamples(data)
+                packet_counter += 1
+
+    #
+    # UDP comms functions, to communicate with the Raspberry Pi Pico W board used in the SDR-TRX.
+    #
+
+    def get_parameter(self, string):
+        string = string + "\n"
+        self.command_sock.sendto(string.encode(), (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        return self.get_argument()
+
+    def set_parameter(self, string, arg):
+        string = string + "," + arg + "\r" + "\n"
+        self.command_sock.sendto(string.encode(), (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        print('arg is: ', arg)
+        temp_arg = self.get_argument()
+        print('temp_arg is: ', temp_arg)
+        return True
+
+    def get_argument(self):
+        self.command_sock.setblocking(True)
+        try:
+            data1 = self.command_sock.recv(1024)
+            print('Received on line 178: ', data1)
+        except Exception as e:
+            print("Exception occurred: {0}".format(e))
+            return -1
+        self.command_sock.setblocking(False)
+        if not data1.startswith(b'OK'):
+            print('Received on line 185: ', data1)
+        # Maybe we didn't catch an OK line?
+        else:
+            while True:
+                try:
+                    data1 = self.command_sock.recv(1024)
+                    print('Received on line 190: ', data1)
+                    break
+                except BlockingIOError:
+                    pass
+        # Check to see if we have a comma in the string. If not, there is no argument.
+        if data1.find(b',') == -1:
+            return -1
+
+        data1 = data1.split(b',')[1].rstrip(b'\r\n')
+        print("data1 =")
+        print(data1)
+
+        # Check for the OK string.  Should we wait for the OK string?
+        try:
+            data2 = self.command_sock.recv(1024)
+            if data2.startswith(b'OK'):  
+                return data2
+        except BlockingIOError:
+            return -1
+
+    # We might want to integrate the UDP reads below into the stuff above.
+
+    #  The code below is to interface WSJT-X with SDR-TRX.
+
+    def encode_ft8(self, msg):
+        try:
+            a77 = self.ft8_encoder.pack(msg, 1)
+            symbols = self.ft8_encoder.make_symbols(a77)
+        except Exception as e:
+            print("FT8 encoder error, check message!")
+            print(f"Exception type: {type(e).__name__}")
+            print(f"Exception message: {str(e)}")
+            symbols = None
+            time.sleep(3)
+        return symbols
+
+    def encode_ft4(self, msg):
+        try:
+            a77 = self.ft4_encoder.pack(msg, 1)
+            symbols = self.ft4_encoder.make_symbols(a77)
+        except:
+            print("FT4 encoder error, check message!")
+            symbols = None
+            time.sleep(3)
+        return symbols
+
+    def load_symbols(self, symbols):
+        print("Load symbols into transmitter..")
+        self.command_sock.sendto(b'm', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        count = 0
+        for symbol in symbols:
+            self.command_sock.sendto(struct.pack('>B', symbol), (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+            count += 1
+            # Wait to avoid Arduino serial buffer overflow.  This may not be needed with the Pico.
+            if count % 50 == 0:
+                time.sleep(0.05)
+        self.command_sock.sendto(b'\0', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        resp = self.command_sock.recv(1)
+        if resp == b'm':
+            print("Load OK")
+        else:
+            print(resp)
+
+    def change_freq(self, new_freq):
+        #global self.tx_freq
+        self.command_sock.setblocking(True)
+        print("Change TX frequency to:", new_freq)
+        self.command_sock.sendto(b'o', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        self.command_sock.sendto(struct.pack('<H', new_freq), (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+        time.sleep(0.05)    
+        resp = self.command_sock.recv(1)
+        print("resp after changing offset is: ", resp)
+        self.command_sock.setblocking(False)
+        if resp == b'o':
+            print("New freq OK")
+            self.tx_freq = new_freq
+
+    def set_mode(self, new_mode):  # New more robust protocol.
+        #global mode
+        if new_mode == 'FT8':
+            self.command_sock.sendto(b'e', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+            time.sleep(.05)
+            resp = self.command_sock.read(1) 
+            print("resp = ", resp)       
+            if resp == b'e':
+                self.mode = new_mode
+                print("Switched to: {0}".format(new_mode)) 
+                self.current_msg = '' 
+                return True
+        elif new_mode == 'FT4':
+            self.command_sock.sendto(b'f', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+            time.sleep(0.05)
+            resp = self.command_sock.recv(1)
+            print("resp = ", resp)       
+            if resp == b'f':
+                self.mode = new_mode
+                self.current_msg = ''
+                print("Switched to: {0}".format(new_mode))
+                return True
+        else:
+            return False
+
+
+    def new_msg(self, msg):
+        #global self.current_msg
+        #global self.mode
+        if msg != self.current_msg:
+            print("Message: {0}".format(msg))
+            if 'FT8' in self.mode:
+                symbols = self.encode_ft8(msg)
+            else:
+                symbols = self.encode_ft4(msg)
+            if symbols.any():
+                # symbols = [kk for kk in range(79)]
+                self.load_symbols(symbols)
+                self.current_msg = msg
+            else:
+                return
+        else:
+            time.sleep(0.005)  #  Do we need this?
+
+    def transmit(self):
+        if False:  # not current_msg:
+            print("No previous message!")
+            time.sleep(1)
+        else:
+            print("TX!")
+            self.command_sock.sendto(b't', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+            # self.tx_now = False
+
+    def check_time_window(self, utc_time):
+        time_window = 15 if 'FT8' in self.mode else 7
+        print("Time window: ", time_window)  # Is correct for FT8 and FT4.
+        rm = utc_time.second % time_window
+        if rm > 1 and rm < time_window - 1:  # Within one second of exact time for FT8.
+            return False
+        else:
+            return True
+
+    # See if HeartBeat will work for this:
+    def HeartBeat(self):  # Should be called at 10 Hz from the main.
+        # Check transmitter is initialized
+        #raise Exception("In HeartBeat")
+    
+        # print("\n\nWait for transmitter ready...")
+        if self.tx_ready_wsjtx == False:
+            if self.tx_ready_wsjtx_sent == False:
+                self.command_sock.sendto(b'r', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+                time.sleep(0.05)
+                self.tx_ready_wsjtx_sent = True
+            while True:
+                try:
+                    x = self.command_sock.recv(1)
+                    if x == b'r':
+                        print("Transmitter ready!")
+                        self.tx_ready_wsjtx = True
+                    break
+                except BlockingIOError:
+                    pass          
+        if self.tx_ready_wsjtx == True:
+            try:
+                fileContent = self.wsjtx_sock.recv(1024)
+            except Exception as e:
+                if e == BlockingIOError:  # This is our way of pollig the socket.
+                    # Maybe use select to stop the errors printing after we quit.
+                    return BaseHardware.HeartBeat(self)
+                else:
+                    # print("Error in HeartBeat")
+                    # print(f"Exception type: {type(e).__name__}")
+                    # print(f"Exception message: {str(e)}")
+                    pass
+            try:
+                NewPacket = WSJTXClass.WSJTX_Packet(fileContent, 0)
+                NewPacket.Decode()
+
+                if NewPacket.PacketType == 1:
+                    # print('New Packet is type 1')
+                    StatusPacket = WSJTXClass.WSJTX_Status(fileContent, NewPacket.index)
+                    StatusPacket.Decode()
+    
+                    # Check TX frequency and update transceiver
+                    new_freq = StatusPacket.TxDF
+                    new_mode = StatusPacket.TxMode.strip()
+                    
+
+                    if new_freq != self.tx_freq:
+                        # time.sleep(0.03)
+                        print('Changing frequency from {0} to {1}'.format(self.tx_freq, new_freq))
+                        try:
+                            self.change_freq(new_freq)
+                        except Exception as e:
+                            print(f"Exception occurred: {e}")
+                        print ('It is now ', self.tx_freq)
+
+                    # print('new_mode', new_mode, ' self.mode is: ', self.mode)
+                    if new_mode != self.mode:  
+                        print("Mode before: {0}".format(self.mode))
+                        self.set_mode(new_mode)
+                        print("New mode after: {0}".format(self.mode))
+
+                    # Check if TX is enabled
+                    if StatusPacket.Transmitting == 1:
+                        # Check time, avoid transmitting out of the time slot
+                        current_time = datetime.datetime.now()
+                        utc_time = current_time.astimezone(datetime.timezone.utc)
+                        self.tx_now = self.check_time_window(utc_time)
+                        if self.tx_now:
+                            self.command_sock.sendto(b'p', (self.PICO_UDP_IP, self.COMMAND_UDP_PORT))
+                        message = StatusPacket.TxMessage
+                        message = message.replace('<', '')
+                        message = message.replace('>', '')
+                        print('Message is: ', message)
+                        self.new_msg(message.strip())
+                        print("Encoded message is: ", self.current_msg)
+                        # print("tx_now is: ", self.tx_now)
+                        if self.tx_now:
+                            print('Transmitting')
+                            self.transmit()
+                        print("Time: {0}:{1}:{2}".format(utc_time.hour, utc_time.minute, utc_time.second))
+                    else:
+                        pass
+                        #  print('Not Transmitting')
+                    return BaseHardware.HeartBeat(self)
+        
+            except Exception as e:
+                # print("Error in HeartBeat")
+                # print(f"Exception type: {type(e).__name__}")
+                # print(f"Exception message: {str(e)}")
+                # traceback.print_exc()
+                return BaseHardware.HeartBeat(self)

setup.py

@@ -4,19 +4,14 @@ import os
 
 # This file is used to build the Linux and Mac versions of Quisk. Windows builds are not included.
 
-# You must define the version here.  A title string including
-# the version will be written to __init__.py and read by quisk.py.
-
-Version = '4.2.28'
-
-fp = open("__init__.py", "w")	# write title string
-fp.write("#Quisk version %s\n" % Version)
-fp.write("from .quisk import main\n")
+fp = open('__init__.py')		# Read in the version
+Version = fp.readline().strip()[15:]
 fp.close()
 
 sources = ['quisk.c', 'sound.c', 'is_key_down.c', 'microphone.c', 'utility.c',
 	'sound_alsa.c', 'sound_pulseaudio.c', 'sound_portaudio.c', 'sound_directx.c', 'sound_wasapi.c',
-	'filter.c', 'extdemod.c', 'freedv.c', 'quisk_wdsp.c', 'ac2yd/remote.c']
+	'filter.c', 'extdemod.c', 'freedv.c', 'quisk_wdsp.c', 'ac2yd/remote.c',
+	'tci.c', 'base64.c', 'handshake.c', 'sha1.c', 'utf8.c', 'ws.c']
 
 # Afedri hardware support added by Alex, Alex@gmail.com
 mAfedri = Extension ('quisk.afedrinet.afedrinet_io',
@@ -39,8 +34,10 @@ if sys.platform == "darwin":	# Build for Macintosh
   libraries = ['portaudio', 'fftw3', 'm']
   if os.path.isdir('/opt/local/include'):	# MacPorts
     base_dir = '/opt/local'
-  elif os.path.isdir('/usr/local/include'):	# HomeBrew
+  elif os.path.isdir('/usr/local/include'):	# HomeBrew on macOS Intel 
     base_dir = '/usr/local'
+  elif os.path.isdir('/opt/homebrew/include'): # HomeBrew on Apple Silicon
+    base_dir = '/opt/homebrew'
   else:						# Regular build?
     base_dir = '/usr'
   if os.path.isfile(base_dir + "/include/pulse/pulseaudio.h"):
@@ -83,9 +80,9 @@ N1MM+ and software that uses Hamlib.
 	url = 'http://james.ahlstrom.name/quisk/',
 	packages = ['quisk', 'quisk.n2adr', 'quisk.softrock', 'quisk.freedvpkg',
 		'quisk.hermes', 'quisk.hiqsdr', 'quisk.afedrinet', 'quisk.soapypkg',
-		'quisk.sdrmicronpkg', 'quisk.perseuspkg', 'quisk.ac2yd'],
+		'quisk.sdrmicronpkg', 'quisk.perseuspkg', 'quisk.ac2yd', 'quisk.multuspkg'],
 	package_dir =  {'quisk' : '.'},
-	package_data = {'' : ['*.txt', '*.html', '*.so', '*.dll']},
+	package_data = {'' : ['*.txt', '*.html', '*.so', '*.dll'], 'quisk' : ['icons/*']},
 	entry_points = {'gui_scripts' : ['quisk = quisk.quisk:main', 'quisk_vna = quisk.quisk_vna:main']},
 	ext_modules = Modules,
 	provides = ['quisk'],

sha1.c

@@ -0,0 +1,389 @@
+/*
+ *  sha1.c
+ *
+ *  Description:
+ *      This file implements the Secure Hashing Algorithm 1 as
+ *      defined in FIPS PUB 180-1 published April 17, 1995.
+ *
+ *      The SHA-1, produces a 160-bit message digest for a given
+ *      data stream.  It should take about 2**n steps to find a
+ *      message with the same digest as a given message and
+ *      2**(n/2) to find any two messages with the same digest,
+ *      when n is the digest size in bits.  Therefore, this
+ *      algorithm can serve as a means of providing a
+ *      "fingerprint" for a message.
+ *
+ *  Portability Issues:
+ *      SHA-1 is defined in terms of 32-bit "words".  This code
+ *      uses <stdint.h> (included via "sha1.h" to define 32 and 8
+ *      bit unsigned integer types.  If your C compiler does not
+ *      support 32 bit unsigned integers, this code is not
+ *      appropriate.
+ *
+ *  Caveats:
+ *      SHA-1 is designed to work with messages less than 2^64 bits
+ *      long.  Although SHA-1 allows a message digest to be generated
+ *      for messages of any number of bits less than 2^64, this
+ *      implementation only works with messages with a length that is
+ *      a multiple of the size of an 8-bit character.
+ *
+ */
+
+#include "sha1.h"
+
+/*
+ *  Define the SHA1 circular left shift macro
+ */
+#define SHA1CircularShift(bits,word) \
+                (((word) << (bits)) | ((word) >> (32-(bits))))
+
+/* Local Function Prototyptes */
+void SHA1PadMessage(SHA1Context *);
+void SHA1ProcessMessageBlock(SHA1Context *);
+
+/*
+ *  SHA1Reset
+ *
+ *  Description:
+ *      This function will initialize the SHA1Context in preparation
+ *      for computing a new SHA1 message digest.
+ *
+ *  Parameters:
+ *      context: [in/out]
+ *          The context to reset.
+ *
+ *  Returns:
+ *      sha Error Code.
+ *
+ */
+int SHA1Reset(SHA1Context *context)
+{
+    if (!context)
+    {
+        return shaNull;
+    }
+
+    context->Length_Low             = 0;
+    context->Length_High            = 0;
+    context->Message_Block_Index    = 0;
+
+    context->Intermediate_Hash[0]   = 0x67452301;
+    context->Intermediate_Hash[1]   = 0xEFCDAB89;
+    context->Intermediate_Hash[2]   = 0x98BADCFE;
+    context->Intermediate_Hash[3]   = 0x10325476;
+    context->Intermediate_Hash[4]   = 0xC3D2E1F0;
+
+    context->Computed   = 0;
+    context->Corrupted  = 0;
+
+    return shaSuccess;
+}
+
+/*
+ *  SHA1Result
+ *
+ *  Description:
+ *      This function will return the 160-bit message digest into the
+ *      Message_Digest array  provided by the caller.
+ *      NOTE: The first octet of hash is stored in the 0th element,
+ *            the last octet of hash in the 19th element.
+ *
+ *  Parameters:
+ *      context: [in/out]
+ *          The context to use to calculate the SHA-1 hash.
+ *      Message_Digest: [out]
+ *          Where the digest is returned.
+ *
+ *  Returns:
+ *      sha Error Code.
+ *
+ */
+int SHA1Result( SHA1Context *context,
+                uint8_t Message_Digest[SHA1HashSize])
+{
+    int i;
+
+    if (!context || !Message_Digest)
+    {
+        return shaNull;
+    }
+
+    if (context->Corrupted)
+    {
+        return context->Corrupted;
+    }
+
+    if (!context->Computed)
+    {
+        SHA1PadMessage(context);
+        for(i=0; i<64; ++i)
+        {
+            /* message may be sensitive, clear it out */
+            context->Message_Block[i] = 0;
+        }
+        context->Length_Low = 0;    /* and clear length */
+        context->Length_High = 0;
+        context->Computed = 1;
+
+    }
+
+    for(i = 0; i < SHA1HashSize; ++i)
+    {
+        Message_Digest[i] = context->Intermediate_Hash[i>>2]
+                            >> 8 * ( 3 - ( i & 0x03 ) );
+    }
+
+    return shaSuccess;
+}
+
+/*
+ *  SHA1Input
+ *
+ *  Description:
+ *      This function accepts an array of octets as the next portion
+ *      of the message.
+ *
+ *  Parameters:
+ *      context: [in/out]
+ *          The SHA context to update
+ *      message_array: [in]
+ *          An array of characters representing the next portion of
+ *          the message.
+ *      length: [in]
+ *          The length of the message in message_array
+ *
+ *  Returns:
+ *      sha Error Code.
+ *
+ */
+int SHA1Input(    SHA1Context    *context,
+                  const uint8_t  *message_array,
+                  unsigned       length)
+{
+    if (!length)
+    {
+        return shaSuccess;
+    }
+
+    if (!context || !message_array)
+    {
+        return shaNull;
+    }
+
+    if (context->Computed)
+    {
+        context->Corrupted = shaStateError;
+
+        return shaStateError;
+    }
+
+    if (context->Corrupted)
+    {
+         return context->Corrupted;
+    }
+    while(length-- && !context->Corrupted)
+    {
+    context->Message_Block[context->Message_Block_Index++] =
+                    (*message_array & 0xFF);
+
+    context->Length_Low += 8;
+    if (context->Length_Low == 0)
+    {
+        context->Length_High++;
+        if (context->Length_High == 0)
+        {
+            /* Message is too long */
+            context->Corrupted = 1;
+        }
+    }
+
+    if (context->Message_Block_Index == 64)
+    {
+        SHA1ProcessMessageBlock(context);
+    }
+
+    message_array++;
+    }
+
+    return shaSuccess;
+}
+
+/*
+ *  SHA1ProcessMessageBlock
+ *
+ *  Description:
+ *      This function will process the next 512 bits of the message
+ *      stored in the Message_Block array.
+ *
+ *  Parameters:
+ *      None.
+ *
+ *  Returns:
+ *      Nothing.
+ *
+ *  Comments:
+
+ *      Many of the variable names in this code, especially the
+ *      single character names, were used because those were the
+ *      names used in the publication.
+ *
+ *
+ */
+void SHA1ProcessMessageBlock(SHA1Context *context)
+{
+    const uint32_t K[] =    {       /* Constants defined in SHA-1   */
+                            0x5A827999,
+                            0x6ED9EBA1,
+                            0x8F1BBCDC,
+                            0xCA62C1D6
+                            };
+    int           t;                 /* Loop counter                */
+    uint32_t      temp;              /* Temporary word value        */
+    uint32_t      W[80];             /* Word sequence               */
+    uint32_t      A, B, C, D, E;     /* Word buffers                */
+
+    /*
+     *  Initialize the first 16 words in the array W
+     */
+    for(t = 0; t < 16; t++)
+    {
+        W[t]  = (uint32_t)context->Message_Block[t * 4] << 24;
+        W[t] |= (uint32_t)context->Message_Block[t * 4 + 1] << 16;
+        W[t] |= (uint32_t)context->Message_Block[t * 4 + 2] << 8;
+        W[t] |= (uint32_t)context->Message_Block[t * 4 + 3];
+    }
+
+    for(t = 16; t < 80; t++)
+    {
+       W[t] = SHA1CircularShift(1,W[t-3] ^ W[t-8] ^ W[t-14] ^ W[t-16]);
+    }
+
+    A = context->Intermediate_Hash[0];
+    B = context->Intermediate_Hash[1];
+    C = context->Intermediate_Hash[2];
+    D = context->Intermediate_Hash[3];
+    E = context->Intermediate_Hash[4];
+
+    for(t = 0; t < 20; t++)
+    {
+        temp =  SHA1CircularShift(5,A) +
+                ((B & C) | ((~B) & D)) + E + W[t] + K[0];
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+
+        B = A;
+        A = temp;
+    }
+
+    for(t = 20; t < 40; t++)
+    {
+        temp = SHA1CircularShift(5,A) + (B ^ C ^ D) + E + W[t] + K[1];
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    for(t = 40; t < 60; t++)
+    {
+        temp = SHA1CircularShift(5,A) +
+               ((B & C) | (B & D) | (C & D)) + E + W[t] + K[2];
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    for(t = 60; t < 80; t++)
+    {
+        temp = SHA1CircularShift(5,A) + (B ^ C ^ D) + E + W[t] + K[3];
+        E = D;
+        D = C;
+        C = SHA1CircularShift(30,B);
+        B = A;
+        A = temp;
+    }
+
+    context->Intermediate_Hash[0] += A;
+    context->Intermediate_Hash[1] += B;
+    context->Intermediate_Hash[2] += C;
+    context->Intermediate_Hash[3] += D;
+    context->Intermediate_Hash[4] += E;
+
+    context->Message_Block_Index = 0;
+}
+
+/*
+ *  SHA1PadMessage
+ *
+
+ *  Description:
+ *      According to the standard, the message must be padded to an even
+ *      512 bits.  The first padding bit must be a '1'.  The last 64
+ *      bits represent the length of the original message.  All bits in
+ *      between should be 0.  This function will pad the message
+ *      according to those rules by filling the Message_Block array
+ *      accordingly.  It will also call the ProcessMessageBlock function
+ *      provided appropriately.  When it returns, it can be assumed that
+ *      the message digest has been computed.
+ *
+ *  Parameters:
+ *      context: [in/out]
+ *          The context to pad
+ *      ProcessMessageBlock: [in]
+ *          The appropriate SHA*ProcessMessageBlock function
+ *  Returns:
+ *      Nothing.
+ *
+ */
+
+void SHA1PadMessage(SHA1Context *context)
+{
+    /*
+     *  Check to see if the current message block is too small to hold
+     *  the initial padding bits and length.  If so, we will pad the
+     *  block, process it, and then continue padding into a second
+     *  block.
+     */
+    if (context->Message_Block_Index > 55)
+    {
+        context->Message_Block[context->Message_Block_Index++] = 0x80;
+        while(context->Message_Block_Index < 64)
+        {
+            context->Message_Block[context->Message_Block_Index++] = 0;
+        }
+
+        SHA1ProcessMessageBlock(context);
+
+        while(context->Message_Block_Index < 56)
+        {
+            context->Message_Block[context->Message_Block_Index++] = 0;
+        }
+    }
+    else
+    {
+        context->Message_Block[context->Message_Block_Index++] = 0x80;
+        while(context->Message_Block_Index < 56)
+        {
+
+            context->Message_Block[context->Message_Block_Index++] = 0;
+        }
+    }
+
+    /*
+     *  Store the message length as the last 8 octets
+     */
+    context->Message_Block[56] = context->Length_High >> 24;
+    context->Message_Block[57] = context->Length_High >> 16;
+    context->Message_Block[58] = context->Length_High >> 8;
+    context->Message_Block[59] = context->Length_High;
+    context->Message_Block[60] = context->Length_Low >> 24;
+    context->Message_Block[61] = context->Length_Low >> 16;
+    context->Message_Block[62] = context->Length_Low >> 8;
+    context->Message_Block[63] = context->Length_Low;
+
+    SHA1ProcessMessageBlock(context);
+}

sha1.h

@@ -0,0 +1,73 @@
+/*
+ *  sha1.h
+ *
+ *  Description:
+ *      This is the header file for code which implements the Secure
+ *      Hashing Algorithm 1 as defined in FIPS PUB 180-1 published
+ *      April 17, 1995.
+ *
+ *      Many of the variable names in this code, especially the
+ *      single character names, were used because those were the names
+ *      used in the publication.
+ *
+ *      Please read the file sha1.c for more information.
+ *
+ */
+
+#ifndef _SHA1_H_
+#define _SHA1_H_
+
+#include <stdint.h>
+/*
+ * If you do not have the ISO standard stdint.h header file, then you
+ * must typdef the following:
+ *    name              meaning
+ *  uint32_t         unsigned 32 bit integer
+ *  uint8_t          unsigned 8 bit integer (i.e., unsigned char)
+ *  int_least16_t    integer of >= 16 bits
+ *
+ */
+
+#ifndef _SHA_enum_
+#define _SHA_enum_
+enum
+{
+    shaSuccess = 0,
+    shaNull,            /* Null pointer parameter */
+    shaInputTooLong,    /* input data too long */
+    shaStateError       /* called Input after Result */
+};
+#endif
+#define SHA1HashSize 20
+
+/*
+ *  This structure will hold context information for the SHA-1
+ *  hashing operation
+ */
+typedef struct SHA1Context
+{
+    uint32_t Intermediate_Hash[SHA1HashSize/4]; /* Message Digest  */
+
+    uint32_t Length_Low;            /* Message length in bits      */
+    uint32_t Length_High;           /* Message length in bits      */
+
+                               /* Index into message block array   */
+    int_least16_t Message_Block_Index;
+    uint8_t Message_Block[64];      /* 512-bit message blocks      */
+
+    int Computed;               /* Is the digest computed?         */
+    int Corrupted;             /* Is the message digest corrupted? */
+} SHA1Context;
+
+/*
+ *  Function Prototypes
+ */
+
+int SHA1Reset(  SHA1Context *);
+int SHA1Input(  SHA1Context *,
+                const uint8_t *,
+                unsigned int);
+int SHA1Result( SHA1Context *,
+                uint8_t Message_Digest[SHA1HashSize]);
+
+#endif

soapypkg/makefile

@@ -1,7 +1,24 @@
+MODULE_NAME := soapy
+OBJDIR := ../build/local
 
-soapy2:
-	python2 setup.py build_ext --force --inplace
+CFLAGS			:= $(shell python3 ../makefil.py --cflags)
+LDFLAGS			:= $(shell python3 ../makefil.py --ldflags)
+EXTENSION_SUFFIX	:= $(shell python3 ../makefil.py --suffix)
+TARGET := $(MODULE_NAME)$(EXTENSION_SUFFIX)
 
-soapy3:
+# Default target: Link the shared library object
+$(TARGET): soapy.o $(OBJDIR)/import_quisk_api.o
+	$(CC) -o $@ soapy.o $(LDFLAGS) -lSoapySDR $(OBJDIR)/import_quisk_api.o
+
+# Compile the C source files into object files
+soapy.o: soapy.c ../quisk.h ../filter.h
+	$(CC) -o $@ -c $< $(CFLAGS) -I..
+
+.PHONY: clean
+clean:
+	-rm -f  *.o 
+	-rm -f  $(MODULE_NAME)*.so
+
+.PHONY: soapy-old
+soapy-old:
 	python3 setup.py build_ext --force --inplace
-

soapypkg/soapy.c

@@ -1,5 +1,6 @@
 #include <Python.h>
 #include <complex.h>
+#include <string.h>
 #include <SoapySDR/Device.h>
 #include <SoapySDR/Formats.h>
 #include <SoapySDR/Version.h>
@@ -223,7 +224,7 @@ static PyObject * open_device(PyObject * self, PyObject * args)
 	else {
 		snprintf(buf128, 128, "SoapySDRDevice_make fail: %s", SoapySDRDevice_lastError());
 	}
-	return PyString_FromString(buf128);
+	return PyUnicode_FromString(buf128);
 }
 
 static void get_direc_len(const char * name, int * direction, int * length)
@@ -254,7 +255,7 @@ static PyObject * get_device_list(PyObject * self, PyObject * args)	// Called fr
 	for (i = 0; i < length; i++) {
 		dict = PyDict_New();
 		for (j = 0; j < results[i].size; j++)
-			PyDict_SetItemString(dict, results[i].keys[j], PyString_FromString(results[i].vals[j]));
+			PyDict_SetItemString(dict, results[i].keys[j], PyUnicode_FromString(results[i].vals[j]));
 		PyList_Append(devices, dict);
 		Py_DECREF(dict);
 	}
@@ -329,7 +330,7 @@ static PyObject * set_parameter(PyObject * self, PyObject * args)	// Called from
 		}
 	}
 	if (msg200[0])
-		return PyString_FromString(msg200);
+		return PyUnicode_FromString(msg200);
 	Py_INCREF (Py_None);
 	return Py_None;
 }
@@ -377,7 +378,7 @@ static PyObject * get_parameter(PyObject * self, PyObject * args)	// Called from
 		pylist = PyList_New(0);
 		names = SoapySDRDevice_listAntennas(sdev, direction, 0, &len_list);
 		for (i = 0; i < len_list; i++) {
-			pyobj = PyString_FromString(names[i]);
+			pyobj = PyUnicode_FromString(names[i]);
 			PyList_Append(pylist, pyobj);
 			Py_DECREF(pyobj);
 		}
@@ -440,7 +441,7 @@ static PyObject * get_parameter(PyObject * self, PyObject * args)	// Called from
 		pylist = PyList_New(0);
 		names = SoapySDRDevice_listGains(sdev, direction, 0, &len_list);
 		for (i = 0; i < len_list; i++) {
-			pyobj = PyString_FromString(names[i]);
+			pyobj = PyUnicode_FromString(names[i]);
 			PyList_Append(pylist, pyobj);
 			Py_DECREF(pyobj);
 		}
@@ -450,7 +451,7 @@ static PyObject * get_parameter(PyObject * self, PyObject * args)	// Called from
 	else if ( ! strncmp(name, "soapy_listGainsValues", length)) {
 		pylist = PyList_New(0);
 		pylst2 = PyList_New(0);		// First element is the total gain
-		pyobj = PyString_FromString("total");
+		pyobj = PyUnicode_FromString("total");
 		PyList_Append(pylst2, pyobj);
 		Py_DECREF(pyobj);
 		range = SoapySDRDevice_getGainRange(sdev, direction, 0);
@@ -460,7 +461,7 @@ static PyObject * get_parameter(PyObject * self, PyObject * args)	// Called from
 		names = SoapySDRDevice_listGains(sdev, direction, 0, &len_list);
 		for (i = 0; i < len_list; i++) {
 			pylst2 = PyList_New(0);
-			pyobj = PyString_FromString(names[i]);
+			pyobj = PyUnicode_FromString(names[i]);
 			PyList_Append(pylst2, pyobj);
 			Py_DECREF(pyobj);
 			range = SoapySDRDevice_getGainElementRange(sdev, direction, 0, names[i]);

softrock/hardware_usb.py

@@ -178,6 +178,8 @@ class Hardware(BaseHardware):
       vfo = self.vfo + int(offset * 1000)	# Convert kHz to Hz
       self.ChangeFrequency(vfo, vfo, 'repeater')
     return False
+  def ImmediateChange(self, name):
+    return
   def OnSpot(self, level):
     pass
   def HeartBeat(self):	# Called at about 10 Hz by the main

sound.c

@@ -4,6 +4,7 @@
 #define PY_SSIZE_T_CLEAN
 #include <Python.h>
 #include <complex.h>
+#include <string.h>
 #include <math.h>
 #include <sys/time.h>
 #include <time.h>
@@ -85,8 +86,10 @@ char * sound_format_names[4] = {"Int32", "Int16", "Float32", "Int24"} ;
 static struct wav_file file_rec_audio, file_rec_samples, file_rec_mic;
 static int close_file_rec;
 
-static double digital_output_level = 0.7;
+double digital_output_level = 0.7;
+static double file_play_level = 1.0;
 static int dc_remove_bw=100;			// bandwidth of DC removal filter
+static int multus_cw_samples = 0;		// always send CW Tx samples to the Multus radio, even in Rx
 
 static ty_sample_start pt_sample_start;
 static ty_sample_stop  pt_sample_stop;
@@ -111,7 +114,7 @@ void quisk_sample_level(const char * msg, complex double * cSamples, int nSample
 			level = d;
 	}
 	if (QuiskTimeSec() - time0 > 0.1) {
-		printf ("sample_level %s: %10.6lf count %8d\n", msg, level / scale, count);
+		printf ("peak sample level %s: %14.6lf count %8d\n", msg, level / scale, count);
 		level = 0;
 		count = 0;
 		time0 = QuiskTimeSec();
@@ -175,9 +178,10 @@ static void correct_sample (struct sound_dev * dev, complex double * cSamples, i
 		for (i = 0; i < nSamples; i++) {
 			re = creal(cSamples[i]);
 			im = cimag(cSamples[i]);
-			re = re * dev->AmPhAAAA;
-			im = re * dev->AmPhCCCC + im * dev->AmPhDDDD;
-			cSamples[i] = re + I * im;
+			cSamples[i] = re * dev->AmPhAAAA + I * (re * dev->AmPhCCCC + im * dev->AmPhDDDD);
+			//re = re * dev->AmPhAAAA;
+			//im = re * dev->AmPhCCCC + im * dev->AmPhDDDD;
+			//cSamples[i] = re + I * im;
 		}
 	}
 }
@@ -730,8 +734,11 @@ void * quisk_make_txIQ(struct sound_dev * dev, int rewind)
 	}
 	// amplitude and phase corrections
 	if (dev->doAmplPhase) {
-		dSamp0 = dSamp0 * dev->AmPhAAAA;
-		dSamp1 = dSamp0 * dev->AmPhCCCC + dSamp1 * dev->AmPhDDDD;
+		//dSamp0 = dSamp0 * dev->AmPhAAAA;
+		//dSamp1 = dSamp0 * dev->AmPhCCCC + dSamp1 * dev->AmPhDDDD;
+		d = dSamp0;
+		dSamp0 = d * dev->AmPhAAAA;
+		dSamp1 = d * dev->AmPhCCCC + dSamp1 * dev->AmPhDDDD;
 	}
 	switch (dev->sound_format) {
 	case Int16:
@@ -836,6 +843,34 @@ int quisk_play_sidetone(struct sound_dev * dev)
 	return 0;
 }
 
+#if 0
+static void testing(double complex * cSamples, int nSamples)
+{
+	int i;
+	static double AvgR=0, AvgQ=0, AvgRQ=0;
+	double d, p, Gain, Phase;
+	time_t tim;
+	static time_t time0;
+
+	for (i = 0; i < nSamples; i++) {
+		d = creal(cSamples[i]);
+		AvgR += d * d;
+		p = cimag(cSamples[i]);
+		AvgQ += p * p;
+		AvgRQ += d * p;
+	}
+	tim = time(NULL);
+	if (tim - time0 >= 2) {
+		time0 = tim;
+		Gain = sqrt(AvgR / AvgQ);
+		Phase = AvgRQ / AvgQ / Gain;
+		Phase = asin(Phase);
+		printf("Time domain Gain %10.7lf, Phase %10.7lf, %10.7lf degrees\n", Gain, Phase, Phase * 57.29578);
+		AvgR = AvgQ = AvgRQ = 0;
+	}
+}
+#endif
+
 int quisk_read_sound(void)	// Called from sound thread
 {  // called in an infinite loop by the main program
 	int i, nSamples, mic_count, mic_interp, retval, is_cw, mic_sample_rate;
@@ -913,9 +948,10 @@ int quisk_read_sound(void)	// Called from sound thread
 	//QuiskPrintTime("quisk_read_sound end", 0);
 		if (Capture.channel_Delay >= 0)	// delay the I or Q channel by one sample
 			delay_sample(&Capture, (double *)cSamples, nSamples);
+		DCremove(cSamples, nSamples, quisk_sound_state.sample_rate, key_state);
+		//testing(cSamples, nSamples);
 		if (Capture.doAmplPhase)		// amplitude and phase corrections
 			correct_sample(&Capture, cSamples, nSamples);
-		DCremove(cSamples, nSamples, quisk_sound_state.sample_rate, key_state);
 		if (nSamples <= 0)
 			QuiskSleepMicrosec(2000);
 #if DEBUG_IO > 1
@@ -951,6 +987,8 @@ int quisk_read_sound(void)	// Called from sound thread
 	// Perhaps replace the samples with samples from a file
 	if (quisk_record_state == FILE_PLAY_SAMPLES)
 		quisk_play_samples(cSamples, nSamples);
+	// Send radio IQ samples to TCI
+	tci_send_iq(cSamples, nSamples);
 #if ! DEBUG_MIC
 	nSamples = quisk_process_samples(cSamples, nSamples);
 #endif
@@ -965,22 +1003,28 @@ int quisk_read_sound(void)	// Called from sound thread
 	if (remote_control_slave)
 		send_remote_radio_sound_socket(cSamples, nSamples);
 
-	is_DGT = rxMode == DGT_U || rxMode == DGT_L || rxMode == DGT_IQ || rxMode == DGT_FM;
+	is_DGT = (rxMode == DGT_U || rxMode == DGT_L || rxMode == DGT_IQ || rxMode == DGT_FM || rxMode == FDV_U || rxMode == FDV_L);
 	if (quisk_record_state == TMP_PLAY_SPKR_MIC)
 		quisk_tmp_playback(cSamples, nSamples, 1.0);		// replace radio sound
 	else if (quisk_record_state == FILE_PLAY_SPKR_MIC)
-		quisk_file_playback(cSamples, nSamples, 1.0);		// replace radio sound
+		quisk_file_playback(cSamples, nSamples, file_play_level);		// replace radio sound
 	// Play the demodulated audio
 #if DEBUG_MIC != 2
-	if ( ! quisk_play_sidetone(&quisk_Playback))	// play sidetone
-		play_sound_interface(&quisk_Playback, nSamples, cSamples, 1, quisk_audioVolume);	// else play radio sound
+	if ( ! quisk_play_sidetone(&quisk_Playback)) {	// play sidetone
+		if (rxMode == FDV_U || rxMode == FDV_L)
+			play_sound_interface(&quisk_Playback, nSamples, cSamples, 1, 0);	// play silence
+		else
+			play_sound_interface(&quisk_Playback, nSamples, cSamples, 1, quisk_audioVolume);	// play radio sound
+	}
 #endif
 	if (radio_sound_socket != INVALID_SOCKET)
 		send_radio_sound_socket(cSamples, nSamples, quisk_audioVolume);
    
 	// Play digital if required
-	if (is_DGT)
-		play_sound_interface(&DigitalOutput, nSamples, cSamples, 1, digital_output_level);
+	//if (is_DGT)
+	play_sound_interface(&DigitalOutput, nSamples, cSamples, 1, digital_output_level);
+	// Send radio sound to TCI
+	tci_send_audio(cSamples, nSamples);
    
 	// Perhaps record the speaker audio to a file
 	if ( ! key_state && file_rec_audio.fp)
@@ -1026,7 +1070,10 @@ int quisk_read_sound(void)	// Called from sound thread
 		for (i = 0; i < mic_count; i++)
 			cSamples[i] = 0;
 	}
-	//quisk_sample_level("read mic or DGT", cSamples, mic_count, CLIP16);
+	// Receive microphone sound from TCI
+	//quisk_sample_level("read mic or DGT", cSamples, mic_count, CLIP32);
+	mic_count = tci_get_mic(cSamples, mic_count);
+	//quisk_sample_level("quisk_tci", cSamples, mic_count, CLIP32);
 	// Perhaps record the microphone audio to the speaker audio file
 	if (key_state && file_rec_audio.fp)
 		quisk_record_audio(&file_rec_audio, cSamples, mic_count);
@@ -1075,7 +1122,7 @@ int quisk_read_sound(void)	// Called from sound thread
 		if (is_cw) {	// Transmit CW; use capture device for timing, not microphone
 			cwCount += (double)retval * quisk_MicPlayback.sample_rate / quisk_sound_state.sample_rate;
 			mic_count = 0;
-			if (QUISK_CWKEY_DOWN || quiskSpotLevel >= 0) {
+			if (QUISK_CWKEY_DOWN || quiskSpotLevel >= 0 || multus_cw_samples) {
 				while (cwCount >= 1.0) {
 					if (cwEnvelope < 1.0) {
 						cwEnvelope += 1. / (quisk_MicPlayback.sample_rate * 5e-3);	// 5 milliseconds
@@ -1401,6 +1448,8 @@ void quisk_open_sound(void)	// Called from GUI thread
 	RawSamplePlayback.channel_I = 0;
 	RawSamplePlayback.channel_Q = 1;
 
+	file_play_level = QuiskGetConfigDouble("file_play_level", 1.0);
+
 	set_num_channels (&Capture);
 	set_num_channels (&quisk_Playback);
 	set_num_channels (&MicCapture);
@@ -1565,10 +1614,9 @@ PyObject * quisk_micplay_channels(PyObject * self, PyObject * args)	// Called fr
 PyObject * quisk_set_sparams(PyObject * self, PyObject * args, PyObject * keywds)
 {  /* Call with keyword arguments ONLY; change local parameters */
 	static char * kwlist[] = {"dc_remove_bw", "digital_output_level", "remote_control_slave", "remote_control_head",
-		NULL} ;
-
-	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|idii", kwlist, &dc_remove_bw, &digital_output_level,
-	&remote_control_slave, &remote_control_head))
+		"file_play_level", "multus_cw_samples", NULL} ;
+	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|idiidi", kwlist, &dc_remove_bw, &digital_output_level,
+	&remote_control_slave, &remote_control_head, &file_play_level, &multus_cw_samples))
 		return NULL;
 	Py_INCREF (Py_None);
 	return Py_None;

sound_alsa.c

@@ -2,6 +2,7 @@
  * This modue provides sound access for QUISK using the ALSA
  * library for Linux.
 */
+#include <string.h>
 #ifdef QUISK_HAVE_ALSA
 #include <Python.h>
 #include <complex.h>
@@ -454,7 +455,7 @@ static int device_list(PyObject * py, snd_pcm_stream_t stream, char * name)
 			}
 			snprintf(buf100, 100, "%s %s (hw:%d,%d)", card_text, pcm_text, card, dev);
 			if (py) {		// add to list of devices
-				PyList_Append(py, PyString_FromString(buf100));
+				PyList_Append(py, PyUnicode_FromString(buf100));
 			}
 			if (name) {		// return the "hw:" name
 				if (strstr(buf100, name)) {
@@ -495,8 +496,37 @@ static snd_pcm_format_t check_formats(struct sound_dev * dev, snd_pcm_hw_params_
 {
 	snd_pcm_format_t format = SND_PCM_FORMAT_UNKNOWN;
 	dev->sample_bytes = 0;
-
+#if 0
+	char * card_name;
+	int card;
+	snd_pcm_info_t * pcm_info;
+	printf("driver %s\n", snd_ctl_card_info_get_driver(info));
+	int loopback = 0;
+	if (snd_pcm_info_malloc(&pcm_info) == 0) {
+		if (snd_pcm_info (dev->handle, pcm_info) == 0) {
+			card = snd_pcm_info_get_card (pcm_info);
+			if (card >= 0) {
+				if (snd_card_get_name(card, &card_name) == 0) {
+					if (strcmp(card_name, "Loopback") == 0)
+						loopback = 1;
+					printf("name %s\n", card_name);
+					free(card_name);
+				}
+			}
+		}
+		snd_pcm_info_free(pcm_info);
+	}
+#endif
 	strMcpy (dev->msg1, "Available formats: ", QUISK_SC_SIZE);
+	if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S16) == 0) {
+		if (!dev->sample_bytes) {
+			strncat(dev->msg1, "*", QUISK_SC_SIZE);
+			dev->sample_bytes = 2;
+			dev->sound_format = Int16;
+			format = SND_PCM_FORMAT_S16;
+		}
+		strncat(dev->msg1, "S16 ", QUISK_SC_SIZE);
+	}
 	if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S32) == 0) {
 		if (!dev->sample_bytes) {
 			strncat(dev->msg1, "*", QUISK_SC_SIZE);
@@ -515,15 +545,6 @@ static snd_pcm_format_t check_formats(struct sound_dev * dev, snd_pcm_hw_params_
 	if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_U24) == 0) {
 		strncat(dev->msg1, "U24 ", QUISK_SC_SIZE);
 	}
-	if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S16) == 0) {
-		if (!dev->sample_bytes) {
-			strncat(dev->msg1, "*", QUISK_SC_SIZE);
-			dev->sample_bytes = 2;
-			dev->sound_format = Int16;
-			format = SND_PCM_FORMAT_S16;
-		}
-		strncat(dev->msg1, "S16 ", QUISK_SC_SIZE);
-	}
 	if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S24_3LE) == 0) {
 		if (!dev->sample_bytes) {
 			strncat(dev->msg1, "*", QUISK_SC_SIZE);
@@ -1109,10 +1130,10 @@ static void midi_in_devices(PyObject * pylist, int just_names)
 					}
 					else {
 						pytup = PyTuple_New(2);
-						PyList_Append(pylist, pytup);
-						PyTuple_SET_ITEM(pytup, 0, PyUnicode_DecodeUTF8(friendly_name, strlen(friendly_name), "replace"));
+						PyTuple_SetItem(pytup, 0, PyUnicode_DecodeUTF8(friendly_name, strlen(friendly_name), "replace"));
 						snprintf(card_name, 32, "hw:%d,%d,%d", card, device, sub);
-						PyTuple_SET_ITEM(pytup, 1, PyUnicode_DecodeUTF8(card_name, strlen(card_name), "replace"));
+						PyTuple_SetItem(pytup, 1, PyUnicode_DecodeUTF8(card_name, strlen(card_name), "replace"));
+						PyList_Append(pylist, pytup);
 					}
 				}
 			}

sound_directx.c

@@ -1,3 +1,5 @@
+#include <string.h>
+
 #ifdef QUISK_HAVE_DIRECTX
 
 #include <Python.h>
@@ -55,14 +57,7 @@ static int match_name(LPCTSTR lpszDesc, const char * name)
 		Py_DECREF(py_unicode);
 		return 0;
 	}
-#if PY_MAJOR_VERSION >= 3
-	if (PyUnicode_READY(py_unicode) == 0)
-		length = PyUnicode_GET_LENGTH(py_unicode);
-	else
-		length = 0;
-#else
-	length = PyUnicode_GET_SIZE(py_unicode);
-#endif
+	length = PyUnicode_GetLength(py_unicode);
 	if (length <= 0) {
 		Py_DECREF(py_unicode);
 		Py_DECREF(py_substring);
@@ -199,14 +194,14 @@ static int quisk_open_capture(struct sound_dev * dev)
 	DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)DsEnumCapture, dev);
 	if (errFound != DS_OK) {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound capture device name %s not found", dev->name);
+			"DirectSound capture device name %.200s not found", dev->name);
 			if (quisk_sound_state.verbose_sound)
 				QuiskPrintf("%s\n", dev->dev_errmsg);
 		return 1;
 	}
 	if (errOpen != DS_OK) {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound capture device %s open failed", dev->name);
+			"DirectSound capture device %.200s open failed", dev->name);
 			if (quisk_sound_state.verbose_sound)
 				QuiskPrintf("%s\n", dev->dev_errmsg);
 		return 1;
@@ -226,7 +221,7 @@ static int quisk_open_capture(struct sound_dev * dev)
 	}
 	else {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound capture device %s buffer create failed (0x%lX)", dev->name, hr);
+			"DirectSound capture device %.200s buffer create failed (0x%lX)", dev->name, hr);
 		return 1;
 	}
 	ptBuf = (LPDIRECTSOUNDCAPTUREBUFFER)dev->buffer;
@@ -235,7 +230,7 @@ static int quisk_open_capture(struct sound_dev * dev)
 		if (quisk_sound_state.verbose_sound)
 			QuiskPrintf("Capture start error 0x%lX", hr);
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound capture device %s capture start failed", dev->name);
+			"DirectSound capture device %.200s capture start failed", dev->name);
 		return 1;
 	}
 	if (quisk_sound_state.verbose_sound) {
@@ -270,18 +265,18 @@ static int quisk_open_playback(struct sound_dev * dev)
 	DirectSoundEnumerate((LPDSENUMCALLBACK)DsEnumPlay, dev);
 	if (errFound != DS_OK) {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound play device name %s not found", dev->name);
+			"DirectSound play device name %.200s not found", dev->name);
 		return 1;
 	}
 	if (errOpen != DS_OK) {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound play device %s open failed", dev->name);
+			"DirectSound play device %.200s open failed", dev->name);
 		return 1;
 	}
 	hr = IDirectSound_SetCooperativeLevel ((LPDIRECTSOUND8)dev->handle, quisk_mainwin_handle, DSSCL_PRIORITY);
 	if (hr != DS_OK) {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound play device %s cooperative level failed", dev->name);
+			"DirectSound play device %.200s cooperative level failed", dev->name);
 		return 1;
 	}
 	dev->sample_bytes = 4;
@@ -299,7 +294,7 @@ static int quisk_open_playback(struct sound_dev * dev)
 	}
 	else {
 		snprintf (dev->dev_errmsg, QUISK_SC_SIZE,
-			"DirectSound play device %s buffer create failed (0x%X)", dev->name, (unsigned int)hr);
+			"DirectSound play device %.180s buffer create failed (0x%X)", dev->name, (unsigned int)hr);
 		return 1;
 	}
 	if (quisk_sound_state.verbose_sound) {

sound_portaudio.c

@@ -1,6 +1,8 @@
 /*
  * This module provides sound access for QUISK using the portaudio library.
 */
+#include <string.h>
+
 #ifdef QUISK_HAVE_PORTAUDIO
 
 #include <Python.h>
@@ -475,7 +477,7 @@ static int device_list(PyObject * py, int input)
                 // found one
                 if (py) {
                     snprintf(buf100, 100, "%s", info->name);
-					PyList_Append(py, PyString_FromString(buf100));
+					PyList_Append(py, PyUnicode_FromString(buf100));
                 } 
             }
         }

sound_pulseaudio.c

@@ -34,6 +34,8 @@
  * 2020 Apr by N2ADR Changes in printf's
 */
 
+#include <string.h>
+
 #ifdef QUISK_HAVE_PULSEAUDIO
 
 #include <Python.h>
@@ -64,12 +66,20 @@ volatile int streams_to_start;
 // remember all open devices for easy cleanup on exit
 static pa_stream *OpenPulseDevices[PA_LIST_SIZE * 2] = {NULL};
 
-static int have_QuiskDigitalInput;	// Do we need to create this null sink?
-static int have_QuiskDigitalOutput;	// Do we need to create this null sink?
-
 static int underrunPlayback;
 static int tlength_bytes;
 
+// list of additional sink devices to add
+struct _NewSinks {
+	char * name;	// name of the new sink
+	int create;	// create the sink if it doesn't exist
+};
+#define NUM_NEW_SINKS	2
+static struct _NewSinks sink_list[NUM_NEW_SINKS] = {
+	{"QuiskDigitalInput", 1},
+	{"QuiskDigitalOutput", 1}
+};
+
 /* This callback happens any time a stream changes state. Here, it's primary used to 
  * tell the quisk thread when streams are ready. 
  */
@@ -1022,20 +1032,20 @@ static void source_sink(const char * name, const char * descr, pa_proplist * pro
 	PyObject * pytup;
 
 	pytup = PyTuple_New(3);
-	PyList_Append(pylist, pytup);
-	PyTuple_SET_ITEM(pytup, 0, PyString_FromString(name));
-	PyTuple_SET_ITEM(pytup, 1, PyString_FromString(descr));
+	PyTuple_SetItem(pytup, 0, PyUnicode_FromString(name));
+	PyTuple_SetItem(pytup, 1, PyUnicode_FromString(descr));
 	value = pa_proplist_gets(props, "device.api");
     
 	if (value && ! strcmp(value, "alsa")) {
 		snprintf(buf300, 300, "%s %s (hw:%s,%s)", pa_proplist_gets(props, "alsa.card_name"), pa_proplist_gets(props, "alsa.name"),
 			 pa_proplist_gets(props, "alsa.card"), pa_proplist_gets(props, "alsa.device"));
 
-		PyTuple_SET_ITEM(pytup, 2, PyString_FromString(buf300));
+		PyTuple_SetItem(pytup, 2, PyUnicode_FromString(buf300));
 	}
 	else {
-		PyTuple_SET_ITEM(pytup, 2, PyString_FromString(""));
+		PyTuple_SetItem(pytup, 2, PyUnicode_FromString(""));
 	}
+	PyList_Append(pylist, pytup);
 }
 
 
@@ -1043,26 +1053,30 @@ static void source_sink(const char * name, const char * descr, pa_proplist * pro
 
 // pa_mainloop will call this function when it's ready to tell us about a sink.
 static void pa_sinklist_cb(pa_context *c, const pa_sink_info *l, int eol, void *userdata) {
-    if (eol > 0)	// If eol is set to a positive number, you're at the end of the list
+	int i;
+
+	if (eol > 0)	// If eol is set to a positive number, you're at the end of the list
 		return;
-    source_sink(l->name, l->description, l->proplist, (PyObject *)userdata);
-    if ( ! strcmp(l->name, "QuiskDigitalInput"))
-        have_QuiskDigitalInput = 1;
-    if ( ! strcmp(l->name, "QuiskDigitalOutput"))
-        have_QuiskDigitalOutput = 1;
+	source_sink(l->name, l->description, l->proplist, (PyObject *)userdata);
+	for (i = 0; i < NUM_NEW_SINKS; i++) {
+		if ( ! strcmp(l->name, sink_list[i].name)) {
+			sink_list[i].create = 0;
+		}
+	}
 }
 
-static void pa_sourcelist_cb(pa_context *c, const pa_source_info *l, int eol, void *userdata) {
-    if (eol > 0)
+static void pa_sourcelist_cb(pa_context *c, const pa_source_info *l, int eol, void *userdata)
+{
+	if (eol > 0)
 		return;
-    source_sink(l->name, l->description, l->proplist, (PyObject *)userdata);
+	source_sink(l->name, l->description, l->proplist, (PyObject *)userdata);
 }
 
 static void index_callback(pa_context *c, uint32_t idx, void *userdata) {
     //printf("%u\n", idx);
 }
 
-PyObject * quisk_pulseaudio_sound_devices(PyObject * self, PyObject * args)
+PyObject * quisk_pulseaudio_sound_devices(PyObject * self, PyObject * args)	// Called from the GUI thread
 {	// Return a list of PulseAudio device names [pycapt, pyplay]
 	PyObject * pylist, * pycapt, * pyplay;
 	pa_mainloop *pa_names_ml;
@@ -1070,6 +1084,9 @@ PyObject * quisk_pulseaudio_sound_devices(PyObject * self, PyObject * args)
 	pa_operation *pa_op=NULL;
 	pa_context *pa_names_ctx;
 	int state = 0;
+	int sink_index = 0;
+	char * sink_name;
+	char buf256[256];
 
 	if (!PyArg_ParseTuple (args, ""))
 		return NULL;
@@ -1113,59 +1130,41 @@ PyObject * quisk_pulseaudio_sound_devices(PyObject * self, PyObject * args)
 			pa_mainloop_iterate(pa_names_ml, 1, NULL);
 			break;
 		case 2:
-			// Now we wait for our operation to complete.  When it's
-			// complete our pa_output_devicelist is filled out, and we move
-			// along to the next state
 			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE) {
 				pa_operation_unref(pa_op);
-				// Now we perform another operation to get the source
-				// (input device) list just like before.
-				pa_op = pa_context_get_source_info_list(pa_names_ctx, pa_sourcelist_cb, pycapt);
-				// Update the state so we know what to do next
 				state++;
 			}
-			pa_mainloop_iterate(pa_names_ml, 1, NULL);
+			else
+				pa_mainloop_iterate(pa_names_ml, 1, NULL);
 			break;
+		// The following loads null sinks for digital input and output if they are not present.
 		case 3:
+			if (sink_index >= NUM_NEW_SINKS) {
+				state = 5;
+				continue;
+			}
+			if (sink_list[sink_index].create) {
+				sink_name = sink_list[sink_index].name;
+				snprintf(buf256, 256, "sink_name=%s sink_properties=device.description=%s", sink_name, sink_name);
+				pa_op = pa_context_load_module(pa_names_ctx, "module-null-sink", buf256, index_callback, NULL);
+				state++;
+			}
+			sink_index++;
+			break;
+		case 4:
 			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE) {
 				pa_operation_unref(pa_op);
-				state = 4;
+				state = 3;
 			}
 			else
 				pa_mainloop_iterate(pa_names_ml, 1, NULL);
 			break;
-		// The following loads modules for digital input and output if they are not present.
-		case 4:
-			if ( ! have_QuiskDigitalInput) {
-				pa_op = pa_context_load_module(pa_names_ctx, "module-null-sink",
-					"sink_name=QuiskDigitalInput sink_properties=device.description=QuiskDigitalInput",
-					index_callback, NULL);
-				state = 5;
-				pa_mainloop_iterate(pa_names_ml, 1, NULL);
-			}
-			else
-				state = 6;
-			break;
 		case 5:
-			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE) {
-				pa_operation_unref(pa_op);
-				state = 6;
-			}
-			else
-				pa_mainloop_iterate(pa_names_ml, 1, NULL);
+			pa_op = pa_context_get_source_info_list(pa_names_ctx, pa_sourcelist_cb, pycapt);
+			state++;
+			pa_mainloop_iterate(pa_names_ml, 1, NULL);
 			break;
 		case 6:
-			if ( ! have_QuiskDigitalOutput) {
-				pa_op = pa_context_load_module(pa_names_ctx, "module-null-sink",
-					"sink_name=QuiskDigitalOutput sink_properties=device.description=QuiskDigitalOutput",
-					index_callback, NULL);
-				state = 7;
-				pa_mainloop_iterate(pa_names_ml, 1, NULL);
-			}
-			else
-				state = 9;
-			break;
-		case 7:
 			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE) {
 				pa_operation_unref(pa_op);
 				state = 9;

sound_wasapi.c

@@ -7,6 +7,9 @@
  * Note that there is NO WARRANTY AT ALL.  USE AT YOUR OWN RISK!!
 
 */
+
+#include <string.h>
+
 #ifdef QUISK_HAVE_WASAPI
 
 #define UNICODE
@@ -1125,10 +1128,10 @@ PyObject * quisk_wasapi_sound_devices(PyObject * self, PyObject * args)		// Call
 		EXIT_ON_ERROR(hr)
 		// data items are (name, id, is_raw)
 		pytup = PyTuple_New(3);
+		PyTuple_SetItem(pytup, 0, PyUnicode_FromWideChar(varName.pwszVal, -1));
+		PyTuple_SetItem(pytup, 1, PyUnicode_FromWideChar(pwszID, -1));
+		PyTuple_SetItem(pytup, 2, PyInt_FromLong(1));
 		PyList_Append(pyplay, pytup);
-		PyTuple_SET_ITEM(pytup, 0, PyUnicode_FromWideChar(varName.pwszVal, -1));
-		PyTuple_SET_ITEM(pytup, 1, PyUnicode_FromWideChar(pwszID, -1));
-		PyTuple_SET_ITEM(pytup, 2, PyInt_FromLong(1));
 		CoTaskMemFree(pwszID);
 		pwszID = NULL;
 		PropVariantClear(&varName);
@@ -1155,10 +1158,10 @@ PyObject * quisk_wasapi_sound_devices(PyObject * self, PyObject * args)		// Call
 		EXIT_ON_ERROR(hr)
 		// data items are (name, id, is_raw)
 		pytup = PyTuple_New(3);
+		PyTuple_SetItem(pytup, 0, PyUnicode_FromWideChar(varName.pwszVal, -1));
+		PyTuple_SetItem(pytup, 1, PyUnicode_FromWideChar(pwszID, -1));
+		PyTuple_SetItem(pytup, 2, PyInt_FromLong(1));
 		PyList_Append(pycapt, pytup);
-		PyTuple_SET_ITEM(pytup, 0, PyUnicode_FromWideChar(varName.pwszVal, -1));
-		PyTuple_SET_ITEM(pytup, 1, PyUnicode_FromWideChar(pwszID, -1));
-		PyTuple_SET_ITEM(pytup, 2, PyInt_FromLong(1));
 		CoTaskMemFree(pwszID);
 		pwszID = NULL;
 		PropVariantClear(&varName);

tci.c

@@ -0,0 +1,970 @@
+#include <Python.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include "ws.h"
+#include <pthread.h>
+#include <complex.h>
+#include <time.h>
+#include "quisk.h"
+#include "filter.h"
+
+#define TCI_STREAM_DATA_BYTES	16384
+#define TCI_RX_BUF_SIZE		1024
+#define TCI_COMMAND_SIZE	64
+#define BAD_TCI_IF		999999999
+
+static int verbose;	// non-zero for verbose log messages
+static int tci_port;
+
+// Implement the ExpertSDR2 Version 1.4 protocol:
+// The Tx audio parameters are copied from the Rx audio stream.
+// The only sample type is FLOAT32 type 3.
+// It is unclear whether Stream.length is the number of floats, or half that for two channels. And version 1.4 lacks Stream.channels.
+// The protocol name for TCI version 1.4 is "protocol:ESDR,1.4;" and the port is 40001.
+// The protocol name for TCI version 2.0 is "protocol:ExpertSDR3,2.0;" and the port is 50001.
+// The TCI modulations are: am, sam, dsb, lsb, usb, cw, nfm, wfm, spec, digl, digu, drm.
+
+// These are TCI parameters:
+static long long quisk_dds;
+static long long client_dds=-1;
+static long long quisk_vfo;
+static long long client_vfo=-1;
+static int quisk_if;
+static int client_if=BAD_TCI_IF;
+static int quisk_trx;
+static int client_trx=-1;
+static int quisk_split_enable;
+static int client_split_enable=-1;
+static char   quisk_modulation[TCI_COMMAND_SIZE];
+static char   client_modulation[TCI_COMMAND_SIZE]={0};
+
+static ws_cli_conn_t tci_clients_list[MAX_CLIENTS];	// list of TCI clients
+static int           tci_clients_count;
+static int tci_started;		// Did we start the TCI server?
+
+// Tx Audio
+ws_cli_conn_t	tci_tx_audio_client;	// single client providing Tx audio
+double		tci_tx_audio_time;	// seconds since the start of Tx audio
+uint64_t	tci_tx_audio_samples;	// number of samples requested since the start of Tx audio
+int		tci_tx_audio_rate;	// samples per second
+int		tci_tx_audio_request;	// the Stream.length to request in TX_CHRONO
+
+static pthread_mutex_t clients_list_mutex = PTHREAD_MUTEX_INITIALIZER;
+static pthread_mutex_t tx_buffer_mutex = PTHREAD_MUTEX_INITIALIZER;
+
+enum StreamType		// ExpertSDR3
+{
+IQ_STREAM = 0,		// Receiver IQ signal stream
+RX_AUDIO_STREAM = 1,	// Receiver audio stream
+TX_AUDIO_STREAM = 2,	// Audio stream for transmitter
+TX_CHRONO = 3,		// Time marker flow for audio signal transmission
+LINEOUT_STREAM = 4	// Linear audio output audio stream
+} ;
+
+enum SampleType		// ExpertSDR3
+{
+TCI_INT16 = 0,	// 16-bit integral character type
+TCI_INT24 = 1,	// 24-bit integral character type
+TCI_INT32 = 2,	// 32-bit integral character type
+TCI_FLOAT32 = 3,	// 32-bit type with a floating point
+} ;
+
+struct ClientData {	// data for each client
+	char msg_buf[TCI_RX_BUF_SIZE];
+	int msg_length;
+	int send_Rx_audio_stream;
+	int audio_stream_samplerate;
+	int audio_stream_sample_type;
+	int audio_stream_channels;
+	int audio_stream_samples;
+	int audio_stream_bytes_per_sample;
+	// Rx stream decimation filters
+	struct quisk_cHB45Filter HalfBand1;
+	struct quisk_cFilter filtDecim3;
+	struct quisk_cFilter filtDecim48to24;
+	// Tx stream interpolation filters
+	struct quisk_cHB45Filter HalfBandTx1;
+	struct quisk_cHB45Filter HalfBandTx2;
+	struct quisk_cFilter filtTxInterp3;
+	int send_IQ_stream;
+	int IQ_stream_samplerate;
+} ;
+
+struct _Stream {
+	uint32_t receiver;	// receiver number
+	uint32_t sample_rate;	// sampling rate
+	uint32_t format;	// sample type determined in SampleType
+	uint32_t codec;		// compression algorithm (not implemented), always 0
+	uint32_t crc;		// checksum (not implemented), always 0
+	int32_t  length;	// number of samples
+	uint32_t type;		// stream type determined in StreamType
+	uint32_t channels;	// number of channels for ExpertSDR3
+	uint32_t reserv[8];	// reserved
+	uint8_t  data[TCI_STREAM_DATA_BYTES];	// samples
+} ;
+
+struct _TxChrono {
+	uint32_t receiver;	// receiver number
+	uint32_t sample_rate;	// sampling rate
+	uint32_t format;	// sample type determined in SampleType
+	uint32_t codec;		// compression algorithm (not implemented), always 0
+	uint32_t crc;		// checksum (not implemented), always 0
+	int32_t  length;	// number of samples
+	uint32_t type;		// stream type determined in StreamType
+	uint32_t channels;	// number of channels for ExpertSDR3
+	uint32_t reserv[8];	// reserved
+} ;
+
+static struct _TxChrono TxChrono = {
+	.receiver = 0,
+	.sample_rate = 48000,
+	.format = TCI_FLOAT32,
+	.codec = 0,
+	.crc = 0,
+	.length = 0,
+	.type = TX_CHRONO,
+	.channels = 2,
+	.reserv = {0, 0, 0, 0, 0, 0, 0, 0}
+} ;
+
+struct _StreamTypes {
+	char * name;
+	time_t when;
+} ;
+
+#define NUM_STREAM_TYPES	5
+struct _StreamTypes StreamTypes[NUM_STREAM_TYPES] = {
+	{"iq       ", 0},
+	{"rx_audio ", 0},
+	{"tx_audio ", 0},
+	{"tx_chrono", 0},
+	{"lineout  ", 0}
+} ;
+
+static int interpolate_tx_audio_stream(complex double * cSamples, int nSamples, struct ClientData * ctx);
+static int decimate_audio_stream(complex double * cSamples, int nSamples, struct ClientData * ctx);
+
+static void PrintStream(const char * msg, size_t size, int always)
+{
+	struct _Stream * pt = (struct _Stream *)msg;
+
+	if (always || time(NULL) - StreamTypes[pt->type].when > 5) {
+		StreamTypes[pt->type].when = time(NULL);
+		if (pt->type == TX_AUDIO_STREAM)
+			QuiskPrintf("TCI     Receiving %slength %5d:", StreamTypes[pt->type].name, (int)size);
+		else
+			QuiskPrintf("TCI Sending %s      length %5d:", StreamTypes[pt->type].name, (int)size);
+		QuiskPrintf("%d %5d %d %d %d %5d %d %d\n", (int)(pt->receiver), (int)(pt->sample_rate), (int)(pt->format), (int)(pt->codec),
+			(int)(pt->crc), (int)(pt->length), (int)(pt->type), (int)(pt->channels));
+	}
+}
+
+static int sendframe_txt(ws_cli_conn_t client, const char * msg)
+{
+	if (verbose)
+		QuiskPrintf("TCI Send text              %s\n", msg);
+	return ws_sendframe_txt(client, msg);
+}
+
+static int sendframe_txt_bcast(uint16_t port, const char * msg)
+{
+	if (verbose)
+		QuiskPrintf("TCI Broadcast text         %s\n", msg);
+	return ws_sendframe_txt_bcast(port, msg);
+}
+
+static size_t sendframe_bin(ws_cli_conn_t client, const char * msg, size_t size)
+{
+	if (verbose)
+		PrintStream(msg, size, 0);
+	return ws_sendframe_bin(client, msg, size);
+}
+
+static int text_message(ws_cli_conn_t client, struct ClientData * ctx)
+{  // This function alters the message ctx->msg_buf.
+	int i;
+	char char_buf[TCI_COMMAND_SIZE];
+	char * saveptr, * command;
+	char * arg1, * arg2, * arg3;
+
+	if (verbose)
+		QuiskPrintf("TCI     Received text      %s\n", ctx->msg_buf);
+		//QuiskPrintf("TCI     Received text      %s %20.6lf\n", ctx->msg_buf, QuiskTimeSec());
+	for (i = 0; ctx->msg_buf[i]; i++) {
+		char_buf[i] = ctx->msg_buf[i];		// make a copy of the command
+		ctx->msg_buf[i] = tolower(ctx->msg_buf[i]);
+	}
+	char_buf[i] = 0;
+	command = strtok_r(ctx->msg_buf, ":", &saveptr);
+	if (command == NULL)
+		return 0;
+	arg1 = strtok_r(NULL, ",;", &saveptr);
+	arg2 = strtok_r(NULL, ",;", &saveptr);
+	arg3 = strtok_r(NULL, ",;", &saveptr);
+	// The command will be broadcast to all clients unless the return value is zero.
+	switch (command[0]) {
+	case 'a':
+		if (strcmp(command, "audio_start") == 0) {
+			pthread_mutex_lock(&clients_list_mutex);
+			for (i = 0; i < tci_clients_count; i++) {
+				if (tci_clients_list[i] == client) {
+					ctx->send_Rx_audio_stream = 1;
+					// Rx stream filters
+					memset(&ctx->HalfBand1,
+						0, sizeof(struct quisk_cHB45Filter));
+					quisk_filt_cInit(&ctx->filtDecim3,
+						quiskFilt144D3Coefs, sizeof(quiskFilt144D3Coefs)/sizeof(double));
+					quisk_filt_cInit(&ctx->filtDecim48to24,
+						quiskFilt48dec24Coefs, sizeof(quiskFilt48dec24Coefs)/sizeof(double));
+
+					break;
+				}
+			}
+			pthread_mutex_unlock(&clients_list_mutex);
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "audio_stop") == 0) {
+			pthread_mutex_lock(&clients_list_mutex);
+			for (i = 0; i < tci_clients_count; i++) {
+				if (tci_clients_list[i] == client) {
+					ctx->send_Rx_audio_stream = 0;
+					break;
+				}
+			}
+			pthread_mutex_unlock(&clients_list_mutex);
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "audio_stream_sample_type") == 0) {
+			if (strcmp(arg1, "float32") == 0) {
+				ctx->audio_stream_sample_type = TCI_FLOAT32;
+				ctx->audio_stream_bytes_per_sample = 4;
+				sendframe_txt(client, char_buf);
+			}
+			return 0;
+		}
+		else if (strcmp(command, "audio_samplerate") == 0) {
+			i = atoi(arg1);
+			if (i != 48000 && i != 24000 && i != 12000 && i != 8000)
+				return 0;
+			ctx->audio_stream_samplerate = i;
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "audio_stream_channels") == 0) {
+			if (strcmp(arg1, "1") == 0)
+				ctx->audio_stream_channels = 1;
+			else if (strcmp(arg1, "2") == 0)
+				ctx->audio_stream_channels = 2;
+			else
+				return 0;
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "audio_stream_samples") == 0) {
+			return 0;	// do not send
+		}
+		break;
+	case 'd':
+		if (strcmp(command, "dds") == 0) {
+			if (arg2) {
+				client_dds = atoll(arg2);
+				// Echo comes from quisk_tci_set_params()
+			}
+			else {
+				snprintf(char_buf, TCI_COMMAND_SIZE, "dds:0,%lld;", quisk_dds);
+				sendframe_txt(client, char_buf);
+			}
+			return 0;
+		}
+		break;
+	case 'i':
+		if (strcmp(command, "iq_start") == 0) {
+			pthread_mutex_lock(&clients_list_mutex);
+			for (i = 0; i < tci_clients_count; i++) {
+				if (tci_clients_list[i] == client) {
+					ctx->send_IQ_stream = 1;
+					break;
+				}
+			}
+			pthread_mutex_unlock(&clients_list_mutex);
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "iq_stop") == 0) {
+			pthread_mutex_lock(&clients_list_mutex);
+			for (i = 0; i < tci_clients_count; i++) {
+				if (tci_clients_list[i] == client) {
+					ctx->send_IQ_stream = 0;
+					break;
+				}
+			}
+			pthread_mutex_unlock(&clients_list_mutex);
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "iq_samplerate") == 0) {
+			ctx->IQ_stream_samplerate = quisk_sound_state.sample_rate;
+			snprintf(char_buf, TCI_COMMAND_SIZE, "iq_samplerate:%d;", quisk_sound_state.sample_rate);
+			sendframe_txt(client, char_buf);
+			return 0;
+		}
+		else if (strcmp(command, "if") == 0) {
+			if (arg3) {
+				client_if = atoi(arg3);
+				// Echo comes from quisk_tci_set_params()	UNIMPLEMENTED
+			}
+			else {
+				snprintf(char_buf, TCI_COMMAND_SIZE, "if:0,0,%d;", quisk_if);
+				sendframe_txt(client, char_buf);
+			}
+			return 0;
+		}
+		break;
+	case 'm':
+		if (strcmp(command, "modulation") == 0) {
+			if (arg2) {
+				strncpy(client_modulation, arg2, TCI_COMMAND_SIZE - 1);
+				// Echo comes from quisk_tci_set_params()
+			}
+			else {
+				snprintf(char_buf, TCI_COMMAND_SIZE, "modulation:0,%.9s;", quisk_modulation);
+				sendframe_txt(client, char_buf);
+			}
+			return 0;
+		}
+		break;
+	case 's':
+		if (strcmp(command, "split_enable") == 0) {
+			if (arg2) {
+				if (strcmp(arg2, "true") == 0)
+					client_split_enable = 1;
+				else
+					client_split_enable = 0;
+				// Echo comes from quisk_tci_set_params()
+			}
+			else {
+				if (quisk_split_enable)
+					sendframe_txt(client, "split_enable:0,true;");
+				else
+					sendframe_txt(client, "split_enable:0,false;");
+			}
+			return 0;
+		}
+		break;
+	case 't':
+		if (strcmp(command, "trx") == 0) {
+			if (arg2) {
+				if (strcmp(arg2, "true") == 0) {
+					if (quisk_trx == 0) {
+						pthread_mutex_lock(&tx_buffer_mutex);
+						client_trx = 1;
+						tci_tx_audio_request = TCI_STREAM_DATA_BYTES /
+							ctx->audio_stream_bytes_per_sample / ctx->audio_stream_channels;
+						CircularBuffer(0, NULL, tci_tx_audio_request * 2, 0);
+						tci_tx_audio_client = client;
+						tci_tx_audio_time = QuiskTimeSec();
+						tci_tx_audio_samples = 0;
+						tci_tx_audio_rate = ctx->audio_stream_samplerate;
+						pthread_mutex_unlock(&tx_buffer_mutex);
+						// Tx stream filters
+						memset(&ctx->HalfBandTx1,
+							0, sizeof(struct quisk_cHB45Filter));
+						memset(&ctx->HalfBandTx2,
+							0, sizeof(struct quisk_cHB45Filter));
+						quisk_filt_cInit(&ctx->filtTxInterp3,
+							quiskAudio24p3Coefs, sizeof(quiskAudio24p3Coefs)/sizeof(double));
+					}
+				}
+				else if (client == tci_tx_audio_client) {
+					client_trx = 0;
+					tci_tx_audio_client = 0;
+				}
+				// Echo comes from quisk_tci_set_params()
+			}
+			else {
+				if (quisk_trx)
+					sendframe_txt(client, "trx:0,true;");
+				else
+					sendframe_txt(client, "trx:0,false;");
+			}
+			return 0;
+		}
+		else if (strcmp(command, "tx_stream_audio_buffering") == 0) {
+			return 0;	// do not send
+		}
+		break;
+	case 'v':
+		if (strcmp(command, "vfo") == 0) {
+			if (arg3) {
+				client_vfo = atoll(arg3);
+				// Echo comes from quisk_tci_set_params()
+			}
+			else {
+				snprintf(char_buf, TCI_COMMAND_SIZE, "vfo:0,0,%lld;", quisk_vfo);
+				sendframe_txt(client, char_buf);
+			}
+			return 0;
+		}
+		break;
+	}
+	return 1;	// send to all clients
+}
+
+static void onopen(ws_cli_conn_t client)
+{
+	char *cli;
+	char command[TCI_COMMAND_SIZE];
+	struct ClientData * ctx;
+
+	// Make parameters for each client:
+	ctx = malloc(sizeof(struct ClientData));
+	ctx->msg_length = 0;
+	ctx->send_Rx_audio_stream = 0;
+	ctx->send_IQ_stream = 0;
+	ctx->audio_stream_samplerate = 48000;
+	ctx->IQ_stream_samplerate = quisk_sound_state.sample_rate;
+	ctx->audio_stream_sample_type = TCI_FLOAT32;
+	ctx->audio_stream_channels = 2;
+	ctx->audio_stream_samples = 0;
+	ctx->audio_stream_bytes_per_sample = 4;
+	ws_set_connection_context(client, ctx);
+
+	pthread_mutex_lock(&clients_list_mutex);
+	tci_clients_list[tci_clients_count++] = client;
+	pthread_mutex_unlock(&clients_list_mutex);
+
+	if (verbose) {
+		cli = ws_getaddress(client);
+		QuiskPrintf("TCI *Connection opened, addr: %s\n", cli);
+	}
+
+	sendframe_txt(client, "protocol:ExpertSDR3,2.0;");
+	//sendframe_txt(client, "vfo_limits:30000,30000000;");  Not known due to transverters
+	int if_limit = quisk_sound_state.sample_rate / 2;
+	snprintf(command, TCI_COMMAND_SIZE, "if_limits:%d,%d;", - if_limit, if_limit);
+	sendframe_txt(client, command);
+	sendframe_txt(client, "trx_count:1;");
+	sendframe_txt(client, "channel_count:1;");
+	sendframe_txt(client, "device:QuiskSDR;");
+	sendframe_txt(client, "receive_only:false;");
+	sendframe_txt(client, "modulations_list:usb,lsb,cw,am,fm,digl,digu;");
+	sendframe_txt(client, "ready;");
+
+	sendframe_txt(client, "start;");
+	sendframe_txt(client, "tx_enable:0,true;");
+	snprintf(command, TCI_COMMAND_SIZE, "dds:0,%lld;", quisk_dds);
+	sendframe_txt(client, command);
+	snprintf(command, TCI_COMMAND_SIZE, "if:0,0,%d;", quisk_if);
+	sendframe_txt(client, command);
+	snprintf(command, TCI_COMMAND_SIZE, "modulation:0,%.9s;", quisk_modulation);
+	sendframe_txt(client, command);
+	snprintf(command, TCI_COMMAND_SIZE, "vfo:0,0,%lld;", quisk_vfo);
+	sendframe_txt(client, command);
+	if (quisk_trx)
+		sendframe_txt(client, "trx:0,true;");
+	else
+		sendframe_txt(client, "trx:0,false;");
+	if (quisk_split_enable)
+		sendframe_txt(client, "split_enable:0,true;");
+	else
+		sendframe_txt(client, "split_enable:0,false;");
+}
+
+static void onclose(ws_cli_conn_t client)
+{
+	int i, count;
+
+	pthread_mutex_lock(&clients_list_mutex);
+	count = tci_clients_count;
+	tci_clients_count = 0;
+	for (i = 0; i < count; i++)
+		if (tci_clients_list[i] != client)
+			tci_clients_list[tci_clients_count++] = tci_clients_list[i];
+	pthread_mutex_unlock(&clients_list_mutex);
+
+	free(ws_get_connection_context(client));
+	if (verbose) {
+		char * cli = ws_getaddress(client);
+		QuiskPrintf("TCI *Connection closed, addr: %s\n", cli);
+	}
+}
+
+static void onmessage(ws_cli_conn_t client, const unsigned char *msg, uint64_t size, int type)
+{
+	char echo_copy[TCI_RX_BUF_SIZE];
+	int n;
+	static complex double * cCopy = NULL;
+	static int cCopy_size = 0;
+	struct ClientData * ctx = ws_get_connection_context(client);
+
+	if (type == WS_FR_OP_TXT) {
+		if (size > TCI_RX_BUF_SIZE / 2)
+			return;		// prevent overflow
+		if (ctx->msg_length + size >= TCI_RX_BUF_SIZE - 2)
+			ctx->msg_length = 0;	// prevent overflow
+		memcpy(ctx->msg_buf + ctx->msg_length, msg, size);	// append msg to buffer
+		ctx->msg_length += size;
+		ctx->msg_buf[ctx->msg_length] = '\0';
+		char * semicolon;
+		char save;
+		while ((semicolon = strchr(ctx->msg_buf, ';')) != NULL) {
+			save = *(semicolon + 1);
+			*(semicolon + 1) = '\0';
+			strncpy(echo_copy, ctx->msg_buf, TCI_RX_BUF_SIZE);
+			if (text_message(client, ctx))		// should we send to all clients?
+				sendframe_txt_bcast(tci_port, echo_copy);
+			*(semicolon + 1) = save;
+			size_t processed_len = (semicolon - ctx->msg_buf) + 1;
+			size_t remaining_len = ctx->msg_length - processed_len;
+			memmove(ctx->msg_buf, ctx->msg_buf + processed_len, remaining_len);
+			ctx->msg_length = remaining_len;
+			ctx->msg_buf[ctx->msg_length] = '\0';
+		}
+	}
+	else if (type == WS_FR_OP_BIN) {
+		struct _Stream * pt = (struct _Stream *)msg;
+		int count = 0;
+		float re, im;
+		int channels = TxChrono.channels;
+		int two_channels = channels == 2;
+		int header = 16 * sizeof(uint32_t);
+		const unsigned char * vpt = msg + header;
+		const unsigned char * vpt_end = vpt + (size - header);
+		const unsigned char * vpt2;
+		const unsigned char * vpt2_end;
+		int count2;
+
+		if (verbose >= 3) {
+			QuiskPrintf("\n\nMSG size %ld length %d\n   0 ", size, pt->length);
+			vpt2 = msg + header;
+			vpt2_end = vpt2 + (size - header);
+			count2 = 0;
+			while (vpt2 < vpt2_end) {
+				re = *(float *)vpt2;
+				vpt2 += sizeof(float);
+				im = *(float *)vpt2;
+				vpt2 += sizeof(float);
+				if (fabsf(re) > 1.1)
+					re = 9;
+				if (fabsf(im) > 1.1)
+					im = 9;
+				count2++;
+				QuiskPrintf("%8.4f %8.4f   ", re, im);
+				if (count2 % 5 == 0)
+					QuiskPrintf("\n%4d ", count2);
+			}
+			QuiskPrintf("\n");
+		}
+
+		if (pt->type == TX_AUDIO_STREAM) {
+			n = pt->length * 7 + 100;	// Make room for 6X interpolation
+			if (cCopy_size < n) {
+				if (cCopy)
+					free(cCopy);
+				cCopy_size = n;
+				cCopy = (complex double *)malloc(cCopy_size * sizeof(complex double));
+			}
+			if (pt->length > 0 && client == tci_tx_audio_client) {
+				n = 0;
+				switch (TxChrono.format) {
+				case TCI_INT16:
+					break;
+				case TCI_FLOAT32:
+					// pt->length is the number of floats
+					while (vpt < vpt_end && count < pt->length) {
+						re = *(float *)vpt;
+						vpt += sizeof(float);
+						count++;
+						if (two_channels) {
+							im = *(float *)vpt;
+							vpt += sizeof(float);
+							count++;
+						}
+						else {
+							im = re;
+						}
+						cCopy[n++] = (re + I * im) * (CLIP32 / 2);
+					}
+					if (verbose && count != pt->length)
+						QuiskPrintf("TCI *TX count %ld %d %d\n", size, count, pt->length);
+					n = interpolate_tx_audio_stream(cCopy, n, ctx);
+					pthread_mutex_lock(&tx_buffer_mutex);
+					CircularBuffer(0, cCopy, 0, n);
+					pthread_mutex_unlock(&tx_buffer_mutex);
+					break;
+				}
+			}
+		}
+		if (verbose)
+			PrintStream((const char *)msg, size, 0);
+
+	}
+
+}
+
+static void tci_startup(void)
+{
+	struct ws_server tci = {
+		.host = NULL,
+		.port = 0,
+		.thread_loop = 1,
+		.timeout_ms = 1000,
+		.evs.onopen = &onopen,
+		.evs.onclose = &onclose,
+		.evs.onmessage = &onmessage
+	};
+	char host32[32];
+
+	tci_clients_count = 0;
+	tci_port = QuiskGetConfigInt("tci_port", 40001);
+	strncpy(host32, QuiskGetConfigString("tci_ip", "127.0.0.1"), 31);
+	tci.host = host32;
+	tci.port = tci_port;
+	ws_socket(&tci);
+	if (verbose)
+		QuiskPrintf("TCI Start server on host %s port %d\n", tci.host, tci.port);
+	return;
+}
+
+static int decimate_audio_stream(complex double * cSamples, int nSamples, struct ClientData * ctx)
+{
+	switch(ctx->audio_stream_samplerate) {		// Starting audio rate is 48000
+		case 48000:
+			break;
+		case 24000:
+			nSamples = quisk_cDecimate(cSamples, nSamples, &ctx->filtDecim48to24, 2);
+			break;
+		case 12000:
+			nSamples = quisk_cDecim2HB45(cSamples, nSamples, &ctx->HalfBand1);
+			nSamples = quisk_cDecimate(cSamples, nSamples, &ctx->filtDecim48to24, 2);
+			break;
+		case 8000:
+			nSamples = quisk_cDecim2HB45(cSamples, nSamples, &ctx->HalfBand1);
+			nSamples = quisk_cDecimate(cSamples, nSamples, &ctx->filtDecim3, 3);
+			break;
+		default:
+			nSamples = 0;
+			break;
+	}
+	return nSamples;
+}
+
+static int interpolate_tx_audio_stream(complex double * cSamples, int nSamples, struct ClientData * ctx)
+{
+	switch(tci_tx_audio_rate) {		// Ending audio rate is 48000
+		case 48000:
+			break;
+		case 24000:
+			nSamples = quisk_cInterp2HB45(cSamples, nSamples, &ctx->HalfBandTx1);
+			break;
+		case 12000:
+			nSamples = quisk_cInterp2HB45(cSamples, nSamples, &ctx->HalfBandTx1);
+			nSamples = quisk_cInterp2HB45(cSamples, nSamples, &ctx->HalfBandTx2);
+			break;
+		case 8000:
+			nSamples = quisk_cInterpolate(cSamples, nSamples, &ctx->filtTxInterp3, 3);
+			nSamples = quisk_cInterp2HB45(cSamples, nSamples, &ctx->HalfBandTx1);
+			break;
+		default:
+			nSamples = 0;
+			break;
+	}
+	return nSamples;
+}
+
+void tci_send_audio(complex double * cSamples, int nSamples)	// called from the sound thread
+{
+	int i, n;
+	struct _Stream stream;
+	struct ClientData * ctx;
+	static complex double * cCopy = NULL;
+	static int cCopy_size = 0;
+	complex double * cAudio;
+
+	if (tci_clients_count <= 0)
+		return;
+
+	if (nSamples <= 0)
+		return;
+
+	pthread_mutex_lock(&clients_list_mutex);
+	for (n = 0; n < tci_clients_count; n++) {
+		ctx = ws_get_connection_context(tci_clients_list[n]);
+		if (ctx->send_Rx_audio_stream == 0)
+			continue;
+		memset(&stream, 0, 16 * sizeof(uint32_t));
+		stream.sample_rate = ctx->audio_stream_samplerate;
+		stream.format = ctx->audio_stream_sample_type;
+		stream.type = RX_AUDIO_STREAM;
+		stream.channels = ctx->audio_stream_channels;
+		int two_channels = stream.channels == 2;
+		int bytes_per_sample = ctx->audio_stream_bytes_per_sample;
+		void * vpt = &stream.data;
+		void * vpt_end = vpt + TCI_STREAM_DATA_BYTES;
+		if (ctx->audio_stream_samplerate != 48000) {	// decimation must not change cSamples
+			if (cCopy_size < nSamples) {
+				if (cCopy)
+					free(cCopy);
+				cCopy_size = nSamples * 2;
+				cCopy = (complex double *)malloc(cCopy_size * sizeof(complex double));
+			}
+			memcpy(cCopy, cSamples, nSamples * sizeof(complex double));
+			nSamples = decimate_audio_stream(cCopy, nSamples, ctx);
+			cAudio = cCopy;
+		}
+		else {
+			cAudio = cSamples;
+		}
+		switch (ctx->audio_stream_sample_type) {
+		case TCI_FLOAT32:
+			stream.length = 0;	// make a frame
+			for (i = 0; i < nSamples; i++) {
+				*(float *)vpt = (float)(creal(cAudio[i]) * (1.0 / 2147483648.0 / 2));
+				vpt += bytes_per_sample;
+				stream.length++;
+				if (two_channels) {
+					*(float *)vpt = (float)(cimag(cAudio[i]) * (1.0 / 2147483648.0 / 2));
+					vpt += bytes_per_sample;
+					stream.length++;
+				}
+				if (vpt >= vpt_end || i == nSamples - 1) {
+					sendframe_bin(tci_clients_list[n], (const char *)&stream,
+						(16 * sizeof(uint32_t) + stream.length * bytes_per_sample));
+					stream.length = 0;
+					vpt = &stream.data;
+				}
+			}
+			break;
+		}
+	}
+	pthread_mutex_unlock(&clients_list_mutex);
+}
+
+void tci_send_iq(complex double * cSamples, int nSamples)	// called from the sound thread
+{
+	int i, n;
+	struct _Stream stream;
+	struct ClientData * ctx;
+
+	if (tci_clients_count <= 0)
+		return;
+
+	if (nSamples <= 0)
+		return;
+
+	pthread_mutex_lock(&clients_list_mutex);
+	for (n = 0; n < tci_clients_count; n++) {
+		ctx = ws_get_connection_context(tci_clients_list[n]);
+		if (ctx->send_IQ_stream == 0)
+			continue;
+		memset(&stream, 0, 16 * sizeof(uint32_t));
+		stream.sample_rate = ctx->IQ_stream_samplerate;
+		stream.format = ctx->audio_stream_sample_type;
+		stream.type = IQ_STREAM;
+		stream.channels = 2;
+		int bytes_per_sample = ctx->audio_stream_bytes_per_sample;
+		void * vpt = &stream.data;
+		void * vpt_end = vpt + TCI_STREAM_DATA_BYTES;
+		int original_length = nSamples;
+		switch (ctx->audio_stream_sample_type) {
+		case TCI_FLOAT32:
+			stream.length = 0;	// make a frame
+			for (i = 0; i < nSamples; i++) {
+				*(float *)vpt = (float)(creal(cSamples[i]) * (1.0 / 2147483648.0 / 2));
+				vpt += bytes_per_sample;
+				stream.length++;
+				*(float *)vpt = (float)(cimag(cSamples[i]) * (1.0 / 2147483648.0 / 2));
+				vpt += bytes_per_sample;
+				stream.length++;
+				if (vpt >= vpt_end || i == nSamples - 1) {
+					sendframe_bin(tci_clients_list[n], (const char *)&stream,
+						(16 * sizeof(uint32_t) + stream.length * bytes_per_sample));
+					original_length -= stream.length / 2;
+					stream.length = 0;
+					vpt = &stream.data;
+				}
+			}
+			break;
+		}
+		if (verbose && original_length != 0)
+			QuiskPrintf("Error in tci_send_iq: Samples remain %d\n", original_length);
+	}
+	pthread_mutex_unlock(&clients_list_mutex);
+}
+
+int tci_get_mic(complex double * cSamples, int mic_count)	// called from the sound thread
+{
+	if (tci_tx_audio_client) {
+		pthread_mutex_lock(&tx_buffer_mutex);	// Copy samples from the Tx buffer
+		int count = CircularBuffer(0, cSamples, mic_count, 0);
+		pthread_mutex_unlock(&tx_buffer_mutex);
+		if (count < mic_count) {
+			if (verbose >= 2)
+				QuiskPrintf("TCI *TX replace mic %d %d\n", count, mic_count);
+			while (count < mic_count)
+				cSamples[count++] = 0;
+		}
+		// Ask for more samples
+		double dtime = QuiskTimeSec();
+		if (tci_tx_audio_samples < (dtime - tci_tx_audio_time) * tci_tx_audio_rate) {
+			TxChrono.length = tci_tx_audio_request;
+			sendframe_bin(tci_tx_audio_client, (const char *)&TxChrono, 16 * sizeof(uint32_t));
+			//tci_tx_audio_samples += tci_tx_audio_request;
+			tci_tx_audio_samples += tci_tx_audio_request / 2;	// Stream.length is floats, not samples
+			if (verbose >= 3)
+				QuiskPrintf("TCI Request more Tx samples\n");
+		}
+	}
+	return mic_count;
+}
+
+PyObject * quisk_tci_set_params(PyObject * self, PyObject * args, PyObject * keywds)	// called from the GUI thread
+{  /* Call with keyword arguments ONLY.
+      Sent from Quisk when parameters change. Broadcast the change to clients.*/
+	static char * kwlist[] = {"start", "close", "verbose", "tci_dds", "tci_if", "tci_vfo", "tci_trx", "tci_split_enable",
+		"tci_modulation", NULL} ;
+	long long new_vfo=-1;
+	long long new_dds=-1;
+	int start=-1, close=-1, new_trx=-1, new_split_enable=-1;
+	int new_if = BAD_TCI_IF;
+	char * mode=NULL;
+	char char_buf[TCI_COMMAND_SIZE];
+
+	if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiiLiLiis", kwlist,
+			&start, &close, &verbose, &new_dds, &new_if, &new_vfo, &new_trx, &new_split_enable,
+			&mode))
+		return NULL;
+	if (start == 1) {
+		tci_started = 1;
+		tci_startup();	// start the TCP server for TCI
+	}
+	if (tci_started == 0) {
+		Py_INCREF (Py_None);
+		return Py_None;
+	}
+	if (close != -1) {
+		sendframe_txt_bcast(tci_port, "stop;");
+	}
+	if (new_dds != -1) {
+		quisk_dds = new_dds;
+		snprintf(char_buf, TCI_COMMAND_SIZE, "dds:0,%lld;", quisk_dds);
+		sendframe_txt_bcast(tci_port, char_buf);
+	}
+	if (new_if != BAD_TCI_IF) {
+		quisk_if = new_if;
+		snprintf(char_buf, TCI_COMMAND_SIZE, "if:0,0,%d;", quisk_if);
+		sendframe_txt_bcast(tci_port, char_buf);
+	}
+	if (new_vfo != -1) {
+		quisk_vfo = new_vfo;
+		snprintf(char_buf, TCI_COMMAND_SIZE, "vfo:0,0,%lld;", quisk_vfo);
+		sendframe_txt_bcast(tci_port, char_buf);
+	}
+	if (new_split_enable != -1) {
+		quisk_split_enable = new_split_enable;
+		if (quisk_split_enable)
+			sendframe_txt_bcast(tci_port, "split_enable:0,true;");
+		else
+			sendframe_txt_bcast(tci_port, "split_enable:0,false;");
+	}
+	if (new_trx != -1) {
+		quisk_trx = new_trx;
+		if (quisk_trx) {
+			sendframe_txt_bcast(tci_port, "trx:0,true;");
+		}
+		else {
+			tci_tx_audio_client = 0;
+			sendframe_txt_bcast(tci_port, "trx:0,false;");
+		}
+	}
+	if (mode) {
+		if (strcmp(mode, "DGT-U") == 0)
+			strcpy(quisk_modulation, "digu");
+		else if (strcmp(mode, "DGT-L") == 0)
+			strcpy(quisk_modulation, "digl");
+		else if (strcmp(mode, "CWL") == 0 || strcmp(mode, "CWU") == 0)
+			strcpy(quisk_modulation, "cw");
+		else if (strcmp(mode, "LSB") == 0)
+			strcpy(quisk_modulation, "lsb");
+		else if (strcmp(mode, "USB") == 0)
+			strcpy(quisk_modulation, "usb");
+		else if (strcmp(mode, "AM") == 0)
+			strcpy(quisk_modulation, "am");
+		else if (strcmp(mode, "FM") == 0)
+			strcpy(quisk_modulation, "fm");
+		else
+			strncpy(quisk_modulation, mode, TCI_COMMAND_SIZE - 1);
+		snprintf(char_buf, TCI_COMMAND_SIZE, "modulation:0,%.9s", quisk_modulation);
+		sendframe_txt_bcast(tci_port, char_buf);
+	}
+	Py_INCREF (Py_None);
+	return Py_None;
+}
+
+PyObject * quisk_tci_get_params(PyObject * self, PyObject * args)	// called from the GUI thread
+{  /* Return client request for a parameter change */
+	const char * name;
+	int ii;
+	long long ll;
+	char char_buf[TCI_COMMAND_SIZE];
+
+	if (!PyArg_ParseTuple (args, "s", &name))
+		return NULL;
+
+	if (strcmp(name, "tci_clients_count") == 0)
+		return PyLong_FromLong(tci_clients_count);
+
+	if (tci_started == 0) {
+	}
+	else if (strcmp(name, "tci_vfo") == 0 && client_vfo >= 0) {
+		ll = client_vfo;
+		client_vfo = -1;
+		return PyLong_FromLongLong(ll);
+	}
+	else if (strcmp(name, "tci_dds") == 0 && client_dds >= 0) {
+		ll = client_dds;
+		client_dds = -1;
+		return PyLong_FromLongLong(ll);
+	}
+	else if (strcmp(name, "tci_if") == 0 && client_if != BAD_TCI_IF) {
+		ii = client_if;
+		client_if = BAD_TCI_IF;
+		return PyLong_FromLong(ii);
+	}
+	else if (strcmp(name, "tci_split_enable") == 0 && client_split_enable >= 0) {
+		ii = client_split_enable;
+		client_split_enable = -1;
+		return PyLong_FromLong(ii);
+	}
+	else if (strcmp(name, "tci_trx") == 0 && client_trx >= 0) {
+		ii = client_trx;
+		client_trx = -1;
+		return PyLong_FromLong(ii);
+	}
+	else if (strcmp(name, "tci_modulation") == 0 && client_modulation[0] != 0) {
+		char_buf[0] = 0;
+		if (strcmp(client_modulation, "digu") == 0)
+			strcpy(char_buf, "DGT-U");
+		else if (strcmp(client_modulation, "digl") == 0)
+			strcpy(char_buf, "DGT-L");
+		else if (strcmp(client_modulation, "cw") == 0)
+			strcpy(char_buf, "CWU");
+		else if (strcmp(client_modulation, "lsb") == 0)
+			strcpy(char_buf, "LSB");
+		else if (strcmp(client_modulation, "usb") == 0)
+			strcpy(char_buf, "USB");
+		else if (strcmp(client_modulation, "am") == 0)
+			strcpy(char_buf, "AM");
+		else if (strcmp(client_modulation, "fm") == 0)
+			strcpy(char_buf, "FM");
+		client_modulation[0] = 0;
+		if (char_buf[0])
+			return PyUnicode_FromString(char_buf);
+	}
+	Py_INCREF (Py_None);
+	return Py_None;
+}

utf8.c

@@ -0,0 +1,89 @@
+/**
+ * Copyright (c) 2008-2009 Bjoern Hoehrmann <bjoern@hoehrmann.de>
+ *
+ * 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.
+ */
+
+/*
+ * Amazing utf8 decoder & validator grabbed from:
+ *   http://bjoern.hoehrmann.de/utf-8/decoder/dfa/
+ *
+ * All rights goes to the original author.
+ */
+
+#include "utf8.h"
+
+static const uint8_t utf8d[] = {
+	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, // 00..1f
+	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, // 20..3f
+	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, // 40..5f
+	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, // 60..7f
+	1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9,9, // 80..9f
+	7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, // a0..bf
+	8,8,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2, // c0..df
+	0xa,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x3,0x4,0x3,0x3, // e0..ef
+	0xb,0x6,0x6,0x6,0x5,0x8,0x8,0x8,0x8,0x8,0x8,0x8,0x8,0x8,0x8,0x8, // f0..ff
+	0x0,0x1,0x2,0x3,0x5,0x8,0x7,0x1,0x1,0x1,0x4,0x6,0x1,0x1,0x1,0x1, // s0..s0
+	1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,0,1,1,1,1,1,0,1,0,1,1,1,1,1,1, // s1..s2
+	1,2,1,1,1,1,1,2,1,2,1,1,1,1,1,1,1,1,1,1,1,1,1,2,1,1,1,1,1,1,1,1, // s3..s4
+	1,2,1,1,1,1,1,1,1,2,1,1,1,1,1,1,1,1,1,1,1,1,1,3,1,3,1,1,1,1,1,1, // s5..s6
+	1,3,1,1,1,1,1,3,1,3,1,1,1,1,1,1,1,3,1,1,1,1,1,1,1,1,1,1,1,1,1,1, // s7..s8
+};
+
+static
+uint32_t decode(uint32_t* state, uint32_t* codep, uint32_t byte) {
+	uint32_t type = utf8d[byte];
+
+	*codep = (*state != UTF8_ACCEPT) ?
+		(byte & 0x3fu) | (*codep << 6) :
+		(0xff >> type) & (byte);
+
+	*state = utf8d[256 + *state*16 + type];
+	return *state;
+}
+
+int is_utf8(uint8_t *s) {
+	uint32_t codepoint, state = 0;
+
+	while (*s)
+		decode(&state, &codepoint, *s++);
+
+	return state == UTF8_ACCEPT;
+}
+
+int is_utf8_len(uint8_t *s, size_t len) {
+	uint32_t codepoint, state = 0;
+	size_t i;
+
+	for (i = 0; i < len; i++)
+		decode(&state, &codepoint, *s++);
+
+	return state == UTF8_ACCEPT;
+}
+
+uint32_t is_utf8_len_state(uint8_t *s, size_t len, uint32_t state) {
+	uint32_t codepoint;
+	size_t i;
+
+	codepoint = 0;
+	for (i = 0; i < len; i++)
+		decode(&state, &codepoint, *s++);
+
+	return state;
+}

utf8.h

@@ -0,0 +1,37 @@
+/**
+ * Copyright (c) 2008-2009 Bjoern Hoehrmann <bjoern@hoehrmann.de>
+ *
+ * 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.
+ */
+
+#ifndef UTF8_DECODE_H
+#define UTF8_DECODE_H
+
+	#include <inttypes.h>
+	#include <stddef.h>
+
+	/* UTF8 return state. */
+	#define UTF8_ACCEPT 0
+	#define UTF8_REJECT 1
+
+	extern int is_utf8(uint8_t* s);
+	extern int is_utf8_len(uint8_t *s, size_t len);
+	extern uint32_t is_utf8_len_state(uint8_t *s, size_t len, uint32_t state);
+
+#endif

utility.c

@@ -6,6 +6,7 @@
 #include <sys/time.h>
 #endif
 #include <complex.h>
+#include <string.h>
 #include "quisk.h"
 
 // Access to config file attributes.
@@ -71,40 +72,27 @@ double QuiskGetConfigDouble(const char * name, double deflt)
 
 char * QuiskGetConfigString(const char * name, char * deflt)
 {  // Return the UTF-8 configuration string. Return deflt for failure.
-  char * res;
-  PyObject * attr;
-#if PY_MAJOR_VERSION < 3
-  static char retbuf[QUISK_SC_SIZE];
-#endif
+	Py_ssize_t size;
 
-  if (!quisk_pyConfig || PyErr_Occurred())
-    return deflt;
-  attr = PyObject_GetAttrString(quisk_pyConfig, name);
-  if (attr) {
-#if PY_MAJOR_VERSION >= 3
-    res = (char *)PyUnicode_AsUTF8(attr);
-#else
-    if (PyUnicode_Check(attr)) {
-      PyObject * pystr = PyUnicode_AsUTF8String(attr);
-      strMcpy(retbuf, PyString_AsString(pystr), QUISK_SC_SIZE);
-      retbuf[QUISK_SC_SIZE - 1] = 0;
-      res = retbuf;
-      Py_DECREF(pystr);
-    }
-    else {
-      res = PyString_AsString(attr);
-    }
-#endif
-    Py_DECREF(attr);
-    if (res)
-      return res;		// success
-    else
-      PyErr_Clear();
-  }
-  else {
-    PyErr_Clear();
-  }
-  return deflt;		// failure
+	if (!quisk_pyConfig || PyErr_Occurred())
+		return deflt;		// failure
+	PyObject * attr = PyObject_GetAttrString(quisk_pyConfig, name);
+	if ( ! attr) {
+		PyErr_Clear();
+		return deflt;		// failure
+	}
+	if ( ! PyUnicode_Check(attr)) {
+		Py_DECREF(attr);
+		return deflt;		// failure
+	}
+	const char * utf8_str = PyUnicode_AsUTF8AndSize(attr, &size);
+	if ( ! utf8_str) {
+		Py_DECREF(attr);
+		PyErr_Clear();
+		return deflt;		// failure
+	}
+	Py_DECREF(attr);
+	return (char *)utf8_str;
 }
 
 double QuiskTimeSec(void)
@@ -321,3 +309,66 @@ PyObject * QuiskPrintf(char * format, ...)
         return NULL;
 }
 #endif
+
+#define MAX_CHANNELS	8
+int CircularBuffer(int channel, complex double * cSamples, int nRead, int nWrite)
+{
+// Initialize with cSamples NULL and nRead equal to the buffer delay in samples.
+// Read with nRead, write with nWrite. Only one can be non-zero for read/write. If both are zero, return nBuf.
+// Return the number of samples read or written.
+	static struct _channel {
+		complex double * cBuf;	// circular sample buffer; save samples until the buffer fills to nDelay
+		int sizeBuf;		// current size of cBuf in complex samples
+		int nDelay;		// number of samples needed to start reading the buffer
+		int nBuf;		// current number of complex samples in cBuf
+		int filling;		// if true, we are filling the buffer and not returning samples
+		int Windex;		// index of write
+		int Rindex;		// index of read
+	} Channels[MAX_CHANNELS] = {0};
+	struct _channel * ptChannel = Channels + channel;
+	int i, num;
+
+	if ( ! cSamples) {	// initialize
+		ptChannel->nDelay = nRead;
+		ptChannel->sizeBuf = nRead * 16;
+		ptChannel->cBuf = (complex double *)realloc(ptChannel->cBuf, ptChannel->sizeBuf * sizeof(complex double));
+		memset(ptChannel->cBuf, 0, ptChannel->sizeBuf * sizeof(complex double));
+		ptChannel->nBuf = 0;
+		ptChannel->Windex = 0;
+		ptChannel->Rindex = 0;
+		ptChannel->filling = 1;
+		return 0;
+	}
+	if ( ! ptChannel->cBuf)
+		return 0;
+	if (nRead == 0 && nWrite == 0)
+		return ptChannel->nBuf;
+	num = 0;	// number of samples read or written
+	if (nWrite > 0) {		// Copy samples from cSamples to cBuf
+		for (i = 0; i < nWrite && ptChannel->nBuf < ptChannel->sizeBuf; i++) {
+			ptChannel->cBuf[ptChannel->Windex++] = cSamples[i];
+			if (ptChannel->Windex >= ptChannel->sizeBuf)
+				ptChannel->Windex = 0;
+			ptChannel->nBuf++;
+			num++;
+		}
+		if (ptChannel->nBuf >= ptChannel->nDelay)
+			ptChannel->filling = 0;
+		//if (num != nWrite)
+		//	QuiskPrintf("Circ nWrite %d %d %d %d\n", num, nWrite, ptChannel->nBuf, ptChannel->sizeBuf);
+	}
+	else if (nRead > 0) {
+		if ( ! ptChannel->filling) {		// Copy samples from cBuf to cSamples
+			for (i = 0; i < nRead && ptChannel->nBuf > 0; i++) {
+				cSamples[i] = ptChannel->cBuf[ptChannel->Rindex++];
+				if (ptChannel->Rindex >= ptChannel->sizeBuf)
+					ptChannel->Rindex = 0;
+				ptChannel->nBuf--;
+				num++;
+			}
+			if (ptChannel->nBuf <= 0)
+				ptChannel->filling = 1;
+		}
+	}
+	return num;
+}

wdsp/COPYING

@@ -0,0 +1,340 @@
+                    GNU GENERAL PUBLIC LICENSE
+                       Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.,
+ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+                            Preamble
+
+  The licenses for most software are designed to take away your
+freedom to share and change it.  By contrast, the GNU General Public
+License is intended to guarantee your freedom to share and change free
+software--to make sure the software is free for all its users.  This
+General Public License applies to most of the Free Software
+Foundation's software and to any other program whose authors commit to
+using it.  (Some other Free Software Foundation software is covered by
+the GNU Lesser General Public License instead.)  You can apply it to
+your programs, too.
+
+  When we speak of free software, we are referring to freedom, not
+price.  Our General Public Licenses are designed to make sure that you
+have the freedom to distribute copies of free software (and charge for
+this service if you wish), that you receive source code or can get it
+if you want it, that you can change the software or use pieces of it
+in new free programs; and that you know you can do these things.
+
+  To protect your rights, we need to make restrictions that forbid
+anyone to deny you these rights or to ask you to surrender the rights.
+These restrictions translate to certain responsibilities for you if you
+distribute copies of the software, or if you modify it.
+
+  For example, if you distribute copies of such a program, whether
+gratis or for a fee, you must give the recipients all the rights that
+you have.  You must make sure that they, too, receive or can get the
+source code.  And you must show them these terms so they know their
+rights.
+
+  We protect your rights with two steps: (1) copyright the software, and
+(2) offer you this license which gives you legal permission to copy,
+distribute and/or modify the software.
+
+  Also, for each author's protection and ours, we want to make certain
+that everyone understands that there is no warranty for this free
+software.  If the software is modified by someone else and passed on, we
+want its recipients to know that what they have is not the original, so
+that any problems introduced by others will not reflect on the original
+authors' reputations.
+
+  Finally, any free program is threatened constantly by software
+patents.  We wish to avoid the danger that redistributors of a free
+program will individually obtain patent licenses, in effect making the
+program proprietary.  To prevent this, we have made it clear that any
+patent must be licensed for everyone's free use or not licensed at all.
+
+  The precise terms and conditions for copying, distribution and
+modification follow.
+
+                    GNU GENERAL PUBLIC LICENSE
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License applies to any program or other work which contains
+a notice placed by the copyright holder saying it may be distributed
+under the terms of this General Public License.  The "Program", below,
+refers to any such program or work, and a "work based on the Program"
+means either the Program or any derivative work under copyright law:
+that is to say, a work containing the Program or a portion of it,
+either verbatim or with modifications and/or translated into another
+language.  (Hereinafter, translation is included without limitation in
+the term "modification".)  Each licensee is addressed as "you".
+
+Activities other than copying, distribution and modification are not
+covered by this License; they are outside its scope.  The act of
+running the Program is not restricted, and the output from the Program
+is covered only if its contents constitute a work based on the
+Program (independent of having been made by running the Program).
+Whether that is true depends on what the Program does.
+
+  1. You may copy and distribute verbatim copies of the Program's
+source code as you receive it, in any medium, provided that you
+conspicuously and appropriately publish on each copy an appropriate
+copyright notice and disclaimer of warranty; keep intact all the
+notices that refer to this License and to the absence of any warranty;
+and give any other recipients of the Program a copy of this License
+along with the Program.
+
+You may charge a fee for the physical act of transferring a copy, and
+you may at your option offer warranty protection in exchange for a fee.
+
+  2. You may modify your copy or copies of the Program or any portion
+of it, thus forming a work based on the Program, and copy and
+distribute such modifications or work under the terms of Section 1
+above, provided that you also meet all of these conditions:
+
+    a) You must cause the modified files to carry prominent notices
+    stating that you changed the files and the date of any change.
+
+    b) You must cause any work that you distribute or publish, that in
+    whole or in part contains or is derived from the Program or any
+    part thereof, to be licensed as a whole at no charge to all third
+    parties under the terms of this License.
+
+    c) If the modified program normally reads commands interactively
+    when run, you must cause it, when started running for such
+    interactive use in the most ordinary way, to print or display an
+    announcement including an appropriate copyright notice and a
+    notice that there is no warranty (or else, saying that you provide
+    a warranty) and that users may redistribute the program under
+    these conditions, and telling the user how to view a copy of this
+    License.  (Exception: if the Program itself is interactive but
+    does not normally print such an announcement, your work based on
+    the Program is not required to print an announcement.)
+
+These requirements apply to the modified work as a whole.  If
+identifiable sections of that work are not derived from the Program,
+and can be reasonably considered independent and separate works in
+themselves, then this License, and its terms, do not apply to those
+sections when you distribute them as separate works.  But when you
+distribute the same sections as part of a whole which is a work based
+on the Program, the distribution of the whole must be on the terms of
+this License, whose permissions for other licensees extend to the
+entire whole, and thus to each and every part regardless of who wrote it.
+
+Thus, it is not the intent of this section to claim rights or contest
+your rights to work written entirely by you; rather, the intent is to
+exercise the right to control the distribution of derivative or
+collective works based on the Program.
+
+In addition, mere aggregation of another work not based on the Program
+with the Program (or with a work based on the Program) on a volume of
+a storage or distribution medium does not bring the other work under
+the scope of this License.
+
+  3. You may copy and distribute the Program (or a work based on it,
+under Section 2) in object code or executable form under the terms of
+Sections 1 and 2 above provided that you also do one of the following:
+
+    a) Accompany it with the complete corresponding machine-readable
+    source code, which must be distributed under the terms of Sections
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+
+    b) Accompany it with a written offer, valid for at least three
+    years, to give any third party, for a charge no more than your
+    cost of physically performing source distribution, a complete
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+
+    c) Accompany it with the information you received as to the offer
+    to distribute corresponding source code.  (This alternative is
+    allowed only for noncommercial distribution and only if you
+    received the program in object code or executable form with such
+    an offer, in accord with Subsection b above.)
+
+The source code for a work means the preferred form of the work for
+making modifications to it.  For an executable work, complete source
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+associated interface definition files, plus the scripts used to
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+operating system on which the executable runs, unless that component
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+
+If distribution of executable or object code is made by offering
+access to copy from a designated place, then offering equivalent
+access to copy the source code from the same place counts as
+distribution of the source code, even though third parties are not
+compelled to copy the source along with the object code.
+
+  4. You may not copy, modify, sublicense, or distribute the Program
+except as expressly provided under this License.  Any attempt
+otherwise to copy, modify, sublicense or distribute the Program is
+void, and will automatically terminate your rights under this License.
+However, parties who have received copies, or rights, from you under
+this License will not have their licenses terminated so long as such
+parties remain in full compliance.
+
+  5. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Program or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Program (or any work based on the
+Program), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Program or works based on it.
+
+  6. Each time you redistribute the Program (or any work based on the
+Program), the recipient automatically receives a license from the
+original licensor to copy, distribute or modify the Program subject to
+these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties to
+this License.
+
+  7. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
+excuse you from the conditions of this License.  If you cannot
+distribute so as to satisfy simultaneously your obligations under this
+License and any other pertinent obligations, then as a consequence you
+may not distribute the Program at all.  For example, if a patent
+license would not permit royalty-free redistribution of the Program by
+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Program.
+
+If any portion of this section is held invalid or unenforceable under
+any particular circumstance, the balance of the section is intended to
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+
+It is not the purpose of this section to induce you to infringe any
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+such claims; this section has the sole purpose of protecting the
+integrity of the free software distribution system, which is
+implemented by public license practices.  Many people have made
+generous contributions to the wide range of software distributed
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+system; it is up to the author/donor to decide if he or she is willing
+to distribute software through any other system and a licensee cannot
+impose that choice.
+
+This section is intended to make thoroughly clear what is believed to
+be a consequence of the rest of this License.
+
+  8. If the distribution and/or use of the Program is restricted in
+certain countries either by patents or by copyrighted interfaces, the
+original copyright holder who places the Program under this License
+may add an explicit geographical distribution limitation excluding
+those countries, so that distribution is permitted only in or among
+countries not thus excluded.  In such case, this License incorporates
+the limitation as if written in the body of this License.
+
+  9. The Free Software Foundation may publish revised and/or new versions
+of the General Public License from time to time.  Such new versions will
+be similar in spirit to the present version, but may differ in detail to
+address new problems or concerns.
+
+Each version is given a distinguishing version number.  If the Program
+specifies a version number of this License which applies to it and "any
+later version", you have the option of following the terms and conditions
+either of that version or of any later version published by the Free
+Software Foundation.  If the Program does not specify a version number of
+this License, you may choose any version ever published by the Free Software
+Foundation.
+
+  10. If you wish to incorporate parts of the Program into other free
+programs whose distribution conditions are different, write to the author
+to ask for permission.  For software which is copyrighted by the Free
+Software Foundation, write to the Free Software Foundation; we sometimes
+make exceptions for this.  Our decision will be guided by the two goals
+of preserving the free status of all derivatives of our free software and
+of promoting the sharing and reuse of software generally.
+
+                            NO WARRANTY
+
+  11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
+FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW.  EXCEPT WHEN
+OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
+PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
+OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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+TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU.  SHOULD THE
+PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
+REPAIR OR CORRECTION.
+
+  12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
+REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
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+TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY
+YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER
+PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGES.
+
+                     END OF TERMS AND CONDITIONS
+
+            How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License along
+    with this program; if not, write to the Free Software Foundation, Inc.,
+    51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+
+Also add information on how to contact you by electronic and paper mail.
+
+If the program is interactive, make it output a short notice like this
+when it starts in an interactive mode:
+
+    Gnomovision version 69, Copyright (C) year name of author
+    Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, the commands you use may
+be called something other than `show w' and `show c'; they could even be
+mouse-clicks or menu items--whatever suits your program.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the program, if
+necessary.  Here is a sample; alter the names:
+
+  Yoyodyne, Inc., hereby disclaims all copyright interest in the program
+  `Gnomovision' (which makes passes at compilers) written by James Hacker.
+
+  <signature of Ty Coon>, 1 April 1989
+  Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs.  If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library.  If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.
+

wdsp/Makefile

@@ -0,0 +1,640 @@
+#
+# Makefile for WDSP library
+# This one creates libwdsp.so, the shared library.
+#
+
+LIBS=-lfftw3 -lpthread
+
+#CFLAGS?= -pthread -O3 -D_GNU_SOURCE -Wno-parentheses
+CFLAGS=-g -fPIC -O3 -D _GNU_SOURCE
+
+FFTWINCLUDE=`pkg-config --cflags fftw3`
+
+COMPILE=$(CC) $(CFLAGS) $(FFTWINCLUDE)
+
+SOURCES= amd.c\
+ammod.c\
+amsq.c\
+analyzer.c\
+anf.c\
+anr.c\
+bandpass.c\
+calcc.c\
+calculus.c\
+cblock.c\
+cfcomp.c\
+cfir.c\
+channel.c\
+compress.c\
+delay.c\
+dexp.c\
+div.c\
+eer.c\
+emnr.c\
+emph.c\
+eq.c\
+fcurve.c\
+fir.c\
+firmin.c\
+fmd.c\
+fmmod.c\
+fmsq.c\
+gain.c\
+gen.c\
+icfir.c\
+iir.c\
+iobuffs.c\
+iqc.c\
+linux_port.c\
+lmath.c\
+main.c\
+meter.c\
+meterlog10.c\
+nbp.c\
+nob.c\
+nobII.c\
+osctrl.c\
+patchpanel.c\
+resample.c\
+rmatch.c\
+RXA.c\
+sender.c\
+shift.c\
+siphon.c\
+slew.c\
+snb.c\
+ssql.c \
+syncbuffs.c\
+TXA.c\
+utilities.c\
+varsamp.c\
+version.c\
+wcpAGC.c\
+wisdom.c\
+zetahat.c
+
+HEADERS=amd.h\
+ammod.h\
+amsq.h\
+analyzer.h\
+anf.h\
+anr.h\
+bandpass.h\
+calcc.h\
+calculus.h\
+cblock.h\
+cfcomp.h\
+cfir.h\
+channel.h\
+comm.h\
+compress.h\
+delay.h\
+dexp.h\
+div.h\
+eer.h\
+emnr.h\
+emph.h\
+eq.h\
+fastmath.h\
+fcurve.h\
+fir.h\
+firmin.h \
+fmd.h\
+fmmod.h\
+fmsq.h\
+gain.h\
+gen.h\
+icfir.h\
+iir.h\
+iobuffs.h\
+iqc.h\
+linux_port.h\
+lmath.h\
+main.h\
+meter.h\
+meterlog10.h\
+nbp.h\
+nob.h\
+nobII.h\
+osctrl.h\
+patchpanel.h\
+resample.h\
+resource.h\
+rmatch.h\
+RXA.h\
+sender.h\
+shift.h\
+siphon.h\
+slew.h\
+snb.h\
+ssql.h \
+syncbuffs.h\
+TXA.h\
+utilities.h\
+wcpAGC.h \
+zetahat.h
+
+OBJS=linux_port.o\
+amd.o\
+ammod.o\
+amsq.o\
+analyzer.o\
+anf.o\
+anr.o\
+bandpass.o\
+calcc.o\
+calculus.o\
+cblock.o\
+cfcomp.o\
+cfir.o\
+channel.o\
+compress.o\
+delay.o\
+dexp.o\
+div.o\
+eer.o\
+emnr.o\
+emph.o\
+eq.o\
+fcurve.o\
+fir.o\
+firmin.o\
+fmd.o\
+fmmod.o\
+fmsq.o\
+gain.o\
+gen.o\
+icfir.o\
+iir.o\
+iobuffs.o\
+iqc.o\
+lmath.o\
+main.o\
+meter.o\
+meterlog10.o\
+nbp.o\
+nob.o\
+nobII.o\
+osctrl.o\
+patchpanel.o\
+resample.o\
+rmatch.o\
+RXA.o\
+sender.o\
+shift.o\
+siphon.o\
+slew.o\
+snb.o\
+ssql.o \
+syncbuffs.o\
+TXA.o\
+utilities.o\
+version.o\
+varsamp.o\
+wcpAGC.o\
+wisdom.o\
+zetahat.o
+
+libwdsp.so: $(OBJS)
+	$(CC) -shared -o libwdsp.so $(OBJS) $(LIBS)
+
+.c.o:
+	$(COMPILE) -c -o $@ $<
+
+clean:
+	-rm -f libwdsp.so *.o
+
+#############################################################################
+#
+# What follows is automatically generated by the "makedepend" program
+#
+#############################################################################
+
+# DO NOT DELETE
+
+RXA.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+RXA.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+RXA.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+RXA.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+RXA.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+RXA.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+RXA.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+TXA.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+TXA.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+TXA.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+TXA.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+TXA.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+TXA.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+TXA.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+amd.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+amd.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+amd.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+amd.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+amd.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+amd.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+amd.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+ammod.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+ammod.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+ammod.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+ammod.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+ammod.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+ammod.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+ammod.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+amsq.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+amsq.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+amsq.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+amsq.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+amsq.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+amsq.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+amsq.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+analyzer.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+analyzer.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+analyzer.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+analyzer.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+analyzer.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+analyzer.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+analyzer.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+analyzer.o: utilities.h
+anf.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+anf.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+anf.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+anf.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+anf.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+anf.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+anf.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+anr.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+anr.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+anr.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+anr.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+anr.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+anr.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+anr.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+bandpass.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+bandpass.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+bandpass.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+bandpass.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+bandpass.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+bandpass.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+bandpass.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+bandpass.o: utilities.h
+calcc.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+calcc.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+calcc.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+calcc.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+calcc.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+calcc.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+calcc.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+cblock.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+cblock.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+cblock.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+cblock.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+cblock.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+cblock.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+cblock.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+cfcomp.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+cfcomp.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+cfcomp.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+cfcomp.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+cfcomp.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+cfcomp.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+cfcomp.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+cfir.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+cfir.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+cfir.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+cfir.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+cfir.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+cfir.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+cfir.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+channel.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+channel.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+channel.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+channel.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+channel.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+channel.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+channel.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+channel.o: utilities.h
+comm.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+comm.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+comm.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+comm.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+comm.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+comm.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+comm.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+compress.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+compress.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+compress.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+compress.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+compress.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
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+resample.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+resample.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+resample.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+resample.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+resample.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+resample.o: utilities.h
+rmatch.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+rmatch.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+rmatch.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+rmatch.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+rmatch.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+rmatch.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+rmatch.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+sender.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+sender.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+sender.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+sender.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+sender.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+sender.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+sender.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+shift.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+shift.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+shift.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+shift.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+shift.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+shift.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+shift.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+siphon.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+siphon.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+siphon.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+siphon.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+siphon.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+siphon.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+siphon.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+slew.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+slew.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+slew.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+slew.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+slew.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+slew.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+slew.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+snb.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+snb.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+snb.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+snb.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+snb.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+snb.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+snb.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+ssql.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+ssql.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+ssql.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+ssql.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+ssql.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+ssql.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+ssql.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+syncbuffs.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+syncbuffs.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+syncbuffs.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+syncbuffs.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+syncbuffs.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+syncbuffs.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h
+syncbuffs.o: RXA.h sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h
+syncbuffs.o: TXA.h utilities.h
+utilities.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+utilities.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+utilities.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+utilities.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+utilities.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+utilities.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h
+utilities.o: RXA.h sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h
+utilities.o: TXA.h utilities.h
+varsamp.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+varsamp.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+varsamp.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+varsamp.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+varsamp.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+varsamp.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+varsamp.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+varsamp.o: utilities.h
+version.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+version.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h
+version.o: channel.h compress.h dexp.h div.h eer.h emnr.h emph.h eq.h
+version.o: fcurve.h fir.h fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h
+version.o: icfir.h iobuffs.h iqc.h main.h meter.h meterlog10.h nbp.h nob.h
+version.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+version.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+version.o: utilities.h
+wcpAGC.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+wcpAGC.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+wcpAGC.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+wcpAGC.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+wcpAGC.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+wcpAGC.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+wcpAGC.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+wisdom.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+wisdom.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+wisdom.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+wisdom.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+wisdom.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+wisdom.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+wisdom.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h

wdsp/README.md

@@ -0,0 +1,4 @@
+# wdsp
+WDSP by Warren Pratt, NR0V
+DSP Library originally written for Windows
+Ported to Linux and Android by John Melton g0orx/n6lyt

wdsp/README.txt

@@ -0,0 +1,7 @@
+To use this Linux library with Quisk, enter "make". It supports the NR2 and SNB features.
+The resulting libwdsp.so file is used in-place. No installation is necessary.
+For Windows, the libwdsp.dll file is included in the quisk directory.
+
+The WDSP library was originally written for Windows by Warren Pratt, NR0V.
+John Melton, g0orx/n6lyt, contributed a Linux port.
+Christian, DL1YCF, contributed further improvements.

wdsp/RXA.c

@@ -0,0 +1,958 @@
+/*  RXA.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2014, 2015, 2016, 2023 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#include "comm.h"
+
+struct _rxa rxa[MAX_CHANNELS];
+
+void create_rxa (int channel)
+{
+    rxa[channel].mode = RXA_LSB;
+    rxa[channel].inbuff  = (double *) malloc0 (1 * ch[channel].dsp_insize  * sizeof (complex));
+    rxa[channel].outbuff = (double *) malloc0 (1 * ch[channel].dsp_outsize * sizeof (complex));
+    rxa[channel].midbuff = (double *) malloc0 (2 * ch[channel].dsp_size    * sizeof (complex));
+
+    // shift to select a slice of spectrum
+    rxa[channel].shift.p = create_shift (
+        1,                                              // run
+        ch[channel].dsp_insize,                         // input buffer size
+        rxa[channel].inbuff,                            // pointer to input buffer
+        rxa[channel].inbuff,                            // pointer to output buffer
+        ch[channel].in_rate,                            // samplerate
+        0.0);                                           // amount to shift (Hz)
+
+    // resample to dsp rate for main processing
+    rxa[channel].rsmpin.p = create_resample (
+        0,                                              // run - will be turned ON below if needed
+        ch[channel].dsp_insize,                         // input buffer size
+        rxa[channel].inbuff,                            // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ch[channel].in_rate,                            // input samplerate
+        ch[channel].dsp_rate,                           // output samplerate
+        0.0,                                            // select cutoff automatically
+        0,                                              // select ncoef automatically
+        1.0);                                           // gain
+
+    // signal generator
+    rxa[channel].gen0.p = create_gen (
+        0,                                              // run
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // input buffer
+        rxa[channel].midbuff,                           // output buffer
+        ch[channel].dsp_rate,                           // sample rate
+        2);                                             // mode
+
+    // adc (input) meter
+    rxa[channel].adcmeter.p = create_meter (
+        1,                                              // run
+        0,                                              // optional pointer to another 'run'
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to buffer
+        ch[channel].dsp_rate,                           // samplerate
+        0.100,                                          // averaging time constant
+        0.100,                                          // peak decay time constant
+        rxa[channel].meter,                             // result vector
+        rxa[channel].pmtupdate,                         // locks for meter access
+        RXA_ADC_AV,                                     // index for average value
+        RXA_ADC_PK,                                     // index for peak value
+        -1,                                             // index for gain value
+        0);                                             // pointer for gain computation
+
+    // notch database
+    rxa[channel].ndb.p = create_notchdb (
+        0,                                              // master run for all nbp's
+        1024);                                          // max number of notches
+
+    // notched bandpass
+    rxa[channel].nbp0.p = create_nbp (
+        1,                                              // run, always runs
+        0,                                              // run the notches
+        0,                                              // position
+        ch[channel].dsp_size,                           // buffer size
+        max(2048, ch[channel].dsp_size),                // number of coefficients
+        0,                                              // minimum phase flag
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        -4150.0,                                        // lower filter frequency
+        -150.0,                                         // upper filter frequency
+        ch[channel].dsp_rate,                           // sample rate
+        0,                                              // wintype
+        1.0,                                            // gain
+        1,                                              // auto-increase notch width
+        1025,                                           // max number of passbands
+        &rxa[channel].ndb.p);                           // addr of database pointer
+
+    // bandpass for snba
+    rxa[channel].bpsnba.p = create_bpsnba (
+        0,                                              // bpsnba run flag
+        0,                                              // run the notches
+        0,                                              // position
+        ch[channel].dsp_size,                           // size
+        max(2048, ch[channel].dsp_size),                // number of filter coefficients
+        0,                                              // minimum phase flag
+        rxa[channel].midbuff,                           // input buffer
+        rxa[channel].midbuff,                           // output buffer
+        ch[channel].dsp_rate,                           // samplerate
+        + 250.0,                                        // abs value of cutoff nearest zero
+        + 5700.0,                                       // abs value of cutoff farthest zero
+        - 5700.0,                                       // current low frequency
+        - 250.0,                                        // current high frequency
+        0,                                              // wintype
+        1.0,                                            // gain
+        1,                                              // auto-increase notch width
+        1025,                                           // max number of passbands
+        &rxa[channel].ndb.p);                           // addr of database pointer
+
+    // send spectrum display
+    rxa[channel].sender.p = create_sender (
+        channel == 0,                                   // run
+        0,                                              // flag
+        0,                                              // mode
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        0,                                              // arg0 <- disp
+        1,                                              // arg1 <- ss
+        0,                                              // arg2 <- LO
+        0);                                             // arg3 <- NOT USED
+
+    // S-meter
+    rxa[channel].smeter.p = create_meter (
+        1,                                              // run
+        0,                                              // optional pointer to another 'run'
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to buffer
+        ch[channel].dsp_rate,                           // samplerate
+        0.100,                                          // averaging time constant
+        0.100,                                          // peak decay time constant
+        rxa[channel].meter,                             // result vector
+        rxa[channel].pmtupdate,                         // locks for meter access
+        RXA_S_AV,                                       // index for average value
+        RXA_S_PK,                                       // index for peak value
+        -1,                                             // index for gain value
+        0);                                             // pointer for gain computation
+
+    // AM squelch
+    rxa[channel].amsq.p = create_amsq (
+        0,                                              // run
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to signal input buffer used by xamsq
+        rxa[channel].midbuff,                           // pointer to signal output buffer used by xamsq
+        rxa[channel].midbuff,                           // pointer to trigger buffer that xamsqcap will capture
+        ch[channel].dsp_rate,                           // sample rate
+        0.010,                                          // time constant for averaging signal level
+        0.070,                                          // signal up transition time
+        0.070,                                          // signal down transition time
+        0.009,                                          // signal level to initiate tail
+        0.010,                                          // signal level to initiate unmute
+        0.000,                                          // minimum tail length
+        1.500,                                          // maximum tail length
+        0.0);                                           // muted gain
+
+    // AM demod
+    rxa[channel].amd.p = create_amd (
+        0,                                              // run - OFF by default
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        0,                                              // mode:  0->AM, 1->SAM
+        1,                                              // levelfade:  0->OFF, 1->ON
+        0,                                              // sideband mode:  0->OFF
+        ch[channel].dsp_rate,                           // sample rate
+        -2000.0,                                        // minimum lock frequency
+        +2000.0,                                        // maximum lock frequency
+        1.0,                                            // zeta
+        250.0,                                          // omegaN
+        0.02,                                           // tauR
+        1.4);                                           // tauI
+
+    // FM demod
+    rxa[channel].fmd.p = create_fmd (
+        0,                                              // run
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ch[channel].dsp_rate,                           // sample rate
+        5000.0,                                         // deviation
+        300.0,                                          // f_low
+        3000.0,                                         // f_high
+        -8000.0,                                        // fmin
+        +8000.0,                                        // fmax
+        1.0,                                            // zeta
+        20000.0,                                        // omegaN
+        0.02,                                           // tau - for dc removal
+        0.5,                                            // audio gain
+        1,                                              // run tone filter
+        254.1,                                          // ctcss frequency
+        max(2048, ch[channel].dsp_size),                // # coefs for de-emphasis filter
+        0,                                              // min phase flag for de-emphasis filter
+        max(2048, ch[channel].dsp_size),                // # coefs for audio cutoff filter
+        0);                                             // min phase flag for audio cutoff filter
+
+    // FM squelch
+    rxa[channel].fmsq.p = create_fmsq (
+        0,                                              // run
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input signal buffer
+        rxa[channel].midbuff,                           // pointer to output signal buffer
+        rxa[channel].fmd.p->audio,                      // pointer to trigger buffer
+        ch[channel].dsp_rate,                           // sample rate
+        5000.0,                                         // cutoff freq for noise filter (Hz)
+        &rxa[channel].fmd.p->pllpole,                   // pointer to pole frequency of the fmd pll (Hz)
+        0.100,                                          // delay time after channel flush
+        0.001,                                          // tau for noise averaging
+        0.100,                                          // tau for long noise averaging
+        0.050,                                          // signal up transition time
+        0.010,                                          // signal down transition time
+        0.750,                                          // noise level to initiate tail
+        0.562,                                          // noise level to initiate unmute
+        0.000,                                          // minimum tail time
+        1.200,                                          // maximum tail time
+        max(2048, ch[channel].dsp_size),                // number of coefficients for noise filter
+        0);                                             // minimum phase flag
+
+    // snba
+    rxa[channel].snba.p = create_snba (
+        0,                                              // run
+        rxa[channel].midbuff,                           // input buffer
+        rxa[channel].midbuff,                           // output buffer
+        ch[channel].dsp_rate,                           // input / output sample rate
+        12000,                                          // internal processing sample rate
+        ch[channel].dsp_size,                           // buffer size
+        4,                                              // overlap factor to use
+        256,                                            // frame size to use; sized for 12K rate
+        64,                                             // asize
+        2,                                              // npasses
+        8.0,                                            // k1
+        20.0,                                           // k2
+        10,                                             // b
+        2,                                              // pre
+        2,                                              // post
+        0.5,                                            // pmultmin
+        200.0,                                          // output resampler low cutoff
+        5400.0);                                        // output resampler high cutoff
+
+    // EQ
+    {
+    double default_F[11] = {0.0,  32.0,  63.0, 125.0, 250.0, 500.0, 1000.0, 2000.0, 4000.0, 8000.0, 16000.0};
+    //double default_G[11] = {0.0, -12.0, -12.0, -12.0,  -1.0,  +1.0,   +4.0,   +9.0,  +12.0,  -10.0,   -10.0};
+    double default_G[11] =   {0.0,   0.0,   0.0,   0.0,   0.0,   0.0,    0.0,    0.0,    0.0,    0.0,     0.0};
+    rxa[channel].eqp.p = create_eqp (
+        0,                                              // run - OFF by default
+        ch[channel].dsp_size,                           // buffer size
+        max(2048, ch[channel].dsp_size),                // number of filter coefficients
+        0,                                              // minimum phase flag
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        10,                                             // number of frequencies
+        default_F,                                      // frequency vector
+        default_G,                                      // gain vector
+        0,                                              // cutoff mode
+        0,                                              // wintype
+        ch[channel].dsp_rate);                          // sample rate
+    }
+
+    // ANF
+    rxa[channel].anf.p = create_anf (
+        0,                                              // run - OFF by default
+        0,                                              // position
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ANF_DLINE_SIZE,                                 // dline_size
+        64,                                             // taps
+        16,                                             // delay
+        0.0001,                                         // two_mu
+        0.1,                                            // gamma
+        1.0,                                            // lidx
+        0.0,                                            // lidx_min
+        200.0,                                          // lidx_max
+        6.25e-12,                                       // ngamma
+        6.25e-10,                                       // den_mult
+        1.0,                                            // lincr
+        3.0);                                           // ldecr
+
+    // ANR
+    rxa[channel].anr.p = create_anr (
+        0,                                              // run - OFF by default
+        0,                                              // position
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ANR_DLINE_SIZE,                                 // dline_size
+        64,                                             // taps
+        16,                                             // delay
+        0.0001,                                         // two_mu
+        0.1,                                            // gamma
+        120.0,                                          // lidx
+        120.0,                                          // lidx_min
+        200.0,                                          // lidx_max
+        0.001,                                          // ngamma
+        6.25e-10,                                       // den_mult
+        1.0,                                            // lincr
+        3.0);                                           // ldecr
+
+
+    // EMNR
+    rxa[channel].emnr.p = create_emnr (
+        0,                                              // run
+        0,                                              // position
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // input buffer
+        rxa[channel].midbuff,                           // output buffer
+        4096,                                           // FFT size
+        4,                                              // overlap
+        ch[channel].dsp_rate,                           // samplerate
+        0,                                              // window type
+        1.0,                                            // gain
+        2,                                              // gain method
+        0,                                              // npe_method
+        1);                                             // ae_run
+
+    // AGC
+    rxa[channel].agc.p = create_wcpagc (
+        1,                                              // run
+        3,                                              // mode
+        1,                                              // peakmode = envelope
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ch[channel].dsp_size,                           // buffer size
+        ch[channel].dsp_rate,                           // sample rate
+        0.001,                                          // tau_attack
+        0.250,                                          // tau_decay
+        4,                                              // n_tau
+        10000.0,                                        // max_gain
+        1.5,                                            // var_gain
+        1000.0,                                         // fixed_gain
+        1.0,                                            // max_input
+        1.0,                                            // out_target
+        0.250,                                          // tau_fast_backaverage
+        0.005,                                          // tau_fast_decay
+        5.0,                                            // pop_ratio
+        1,                                              // hang_enable
+        0.500,                                          // tau_hang_backmult
+        0.250,                                          // hangtime
+        0.250,                                          // hang_thresh
+        0.100);                                         // tau_hang_decay
+
+    // agc gain meter
+    rxa[channel].agcmeter.p = create_meter (
+        1,                                              // run
+        0,                                              // optional pointer to another 'run'
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to buffer
+        ch[channel].dsp_rate,                           // samplerate
+        0.100,                                          // averaging time constant
+        0.100,                                          // peak decay time constant
+        rxa[channel].meter,                             // result vector
+        rxa[channel].pmtupdate,                         // locks for meter access
+        RXA_AGC_AV,                                     // index for average value
+        RXA_AGC_PK,                                     // index for peak value
+        RXA_AGC_GAIN,                                   // index for gain value
+        &rxa[channel].agc.p->gain);                     // pointer for gain computation
+
+    // bandpass filter
+    rxa[channel].bp1.p = create_bandpass (
+        1,                                              // run - used only with ( AM || ANF || ANR || EMNR)
+        0,                                              // position
+        ch[channel].dsp_size,                           // buffer size
+        max(2048, ch[channel].dsp_size),                // number of coefficients
+        0,                                              // flag for minimum phase
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        -4150.0,                                        // lower filter frequency
+        -150.0,                                         // upper filter frequency
+        ch[channel].dsp_rate,                           // sample rate
+        1,                                              // wintype
+        1.0);                                           // gain
+
+    // pull phase & scope display data
+    rxa[channel].sip1.p = create_siphon (
+        1,                                              // run - needed only for phase display
+        0,                                              // position
+        0,                                              // mode
+        0,                                              // disp
+        ch[channel].dsp_size,                           // size of input buffer
+        rxa[channel].midbuff,                           // input buffer
+        4096,                                           // number of samples to store
+        4096,                                           // fft size for spectrum
+        0);                                             // specmode
+
+    // carrier block
+    rxa[channel].cbl.p = create_cbl (
+        0,                                              // run - needed only if set to ON
+        ch[channel].dsp_size,                           // buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        0,                                              // mode
+        ch[channel].dsp_rate,                           // sample rate
+        0.02);                                          // tau
+
+    // peaking filter
+    rxa[channel].speak.p = create_speak (
+        0,                                              // run
+        ch[channel].dsp_size,                           // buffer size,
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ch[channel].dsp_rate,                           // sample rate
+        600.0,                                          // center frequency
+        100.0,                                          // bandwidth
+        2.0,                                            // gain
+        4,                                              // number of stages
+        1);                                             // design
+
+    // multiple peak filter
+    {
+        int def_enable[2] = {1, 1};
+        double def_freq[2] = {2125.0, 2295.0};
+        double def_bw[2] = {75.0, 75.0};
+        double def_gain[2] = {1.0, 1.0};
+        rxa[channel].mpeak.p = create_mpeak (
+            0,                                          // run
+            ch[channel].dsp_size,                       // size
+            rxa[channel].midbuff,                       // pointer to input buffer
+            rxa[channel].midbuff,                       // pointer to output buffer
+            ch[channel].dsp_rate,                       // sample rate
+            2,                                          // number of peaking filters
+            def_enable,                                 // enable vector
+            def_freq,                                   // frequency vector
+            def_bw,                                     // bandwidth vector
+            def_gain,                                   // gain vector
+            4 );                                        // number of stages
+    }
+
+    // syllabic squelch
+    rxa[channel].ssql.p = create_ssql(
+        0,                                              // run
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        ch[channel].dsp_rate,                           // sample rate
+        0.070,                                          // signal up transition time
+        0.070,                                          // signal down transition time
+        0.0,                                            // muted gain
+        0.1,                                            // mute time-constant
+        0.1,                                            // unmute time-constant
+        0.08,                                           // window threshold
+        0.8197,                                         // trigger threshold
+        2400,                                           // ring size for f_to_v converter
+        2000.0);                                        // max freq for f_to_v converter
+
+    // patchpanel
+    rxa[channel].panel.p = create_panel (
+        channel,                                        // channel number
+        1,                                              // run
+        ch[channel].dsp_size,                           // size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].midbuff,                           // pointer to output buffer
+        4.0,                                            // gain1
+        1.0,                                            // gain2I
+        1.0,                                            // gain2Q
+        3,                                              // 3 for I and Q
+        0);                                             // no copy
+
+    // resample
+    rxa[channel].rsmpout.p = create_resample (
+        0,                                              // run - will be turned ON below if needed
+        ch[channel].dsp_size,                           // input buffer size
+        rxa[channel].midbuff,                           // pointer to input buffer
+        rxa[channel].outbuff,                           // pointer to output buffer
+        ch[channel].dsp_rate,                           // input sample rate
+        ch[channel].out_rate,                           // output sample rate
+        0.0,                                            // select cutoff automatically
+        0,                                              // select ncoef automatically
+        1.0);                                           // gain
+
+    // turn OFF / ON resamplers as needed
+    RXAResCheck (channel);
+}
+
+void destroy_rxa (int channel)
+{
+    destroy_resample (rxa[channel].rsmpout.p);
+    destroy_panel (rxa[channel].panel.p);
+    destroy_ssql (rxa[channel].ssql.p);
+    destroy_mpeak (rxa[channel].mpeak.p);
+    destroy_speak (rxa[channel].speak.p);
+    destroy_cbl (rxa[channel].cbl.p);
+    destroy_siphon (rxa[channel].sip1.p);
+    destroy_bandpass (rxa[channel].bp1.p);
+    destroy_meter (rxa[channel].agcmeter.p);
+    destroy_wcpagc (rxa[channel].agc.p);
+    destroy_emnr (rxa[channel].emnr.p);
+    destroy_anr (rxa[channel].anr.p);
+    destroy_anf (rxa[channel].anf.p);
+    destroy_eqp (rxa[channel].eqp.p);
+    destroy_snba (rxa[channel].snba.p);
+    destroy_fmsq (rxa[channel].fmsq.p);
+    destroy_fmd (rxa[channel].fmd.p);
+    destroy_amd (rxa[channel].amd.p);
+    destroy_amsq (rxa[channel].amsq.p);
+    destroy_meter (rxa[channel].smeter.p);
+    destroy_sender (rxa[channel].sender.p);
+    destroy_bpsnba (rxa[channel].bpsnba.p);
+    destroy_nbp (rxa[channel].nbp0.p);
+    destroy_notchdb (rxa[channel].ndb.p);
+    destroy_meter (rxa[channel].adcmeter.p);
+    destroy_gen (rxa[channel].gen0.p);
+    destroy_resample (rxa[channel].rsmpin.p);
+    destroy_shift (rxa[channel].shift.p);
+    _aligned_free (rxa[channel].midbuff);
+    _aligned_free (rxa[channel].outbuff);
+    _aligned_free (rxa[channel].inbuff);
+}
+
+void flush_rxa (int channel)
+{
+    memset (rxa[channel].inbuff,  0, 1 * ch[channel].dsp_insize  * sizeof (complex));
+    memset (rxa[channel].outbuff, 0, 1 * ch[channel].dsp_outsize * sizeof (complex));
+    memset (rxa[channel].midbuff, 0, 2 * ch[channel].dsp_size    * sizeof (complex));
+    flush_shift (rxa[channel].shift.p);
+    flush_resample (rxa[channel].rsmpin.p);
+    flush_gen (rxa[channel].gen0.p);
+    flush_meter (rxa[channel].adcmeter.p);
+    flush_nbp (rxa[channel].nbp0.p);
+    flush_bpsnba (rxa[channel].bpsnba.p);
+    flush_sender (rxa[channel].sender.p);
+    flush_meter (rxa[channel].smeter.p);
+    flush_amsq (rxa[channel].amsq.p);
+    flush_amd (rxa[channel].amd.p);
+    flush_fmd (rxa[channel].fmd.p);
+    flush_fmsq (rxa[channel].fmsq.p);
+    flush_snba (rxa[channel].snba.p);
+    flush_eqp (rxa[channel].eqp.p);
+    flush_anf (rxa[channel].anf.p);
+    flush_anr (rxa[channel].anr.p);
+    flush_emnr (rxa[channel].emnr.p);
+    flush_wcpagc (rxa[channel].agc.p);
+    flush_meter (rxa[channel].agcmeter.p);
+    flush_bandpass (rxa[channel].bp1.p);
+    flush_siphon (rxa[channel].sip1.p);
+    flush_cbl (rxa[channel].cbl.p);
+    flush_speak (rxa[channel].speak.p);
+    flush_mpeak (rxa[channel].mpeak.p);
+    flush_ssql (rxa[channel].ssql.p);
+    flush_panel (rxa[channel].panel.p);
+    flush_resample (rxa[channel].rsmpout.p);
+}
+
+void xrxa (int channel)
+{
+    xshift (rxa[channel].shift.p);
+    xresample (rxa[channel].rsmpin.p);
+    xgen (rxa[channel].gen0.p);
+    xmeter (rxa[channel].adcmeter.p);
+    xbpsnbain (rxa[channel].bpsnba.p, 0);
+    xnbp (rxa[channel].nbp0.p, 0);
+    xmeter (rxa[channel].smeter.p);
+    xsender (rxa[channel].sender.p);
+    xamsqcap (rxa[channel].amsq.p);
+    xbpsnbaout (rxa[channel].bpsnba.p, 0);
+    xamd (rxa[channel].amd.p);
+    xfmd (rxa[channel].fmd.p);
+    xfmsq (rxa[channel].fmsq.p);
+    xbpsnbain (rxa[channel].bpsnba.p, 1);
+    xbpsnbaout (rxa[channel].bpsnba.p, 1);
+    xsnba (rxa[channel].snba.p);
+    xeqp (rxa[channel].eqp.p);
+    xanf (rxa[channel].anf.p, 0);
+    xanr (rxa[channel].anr.p, 0);
+    xemnr (rxa[channel].emnr.p, 0);
+    xbandpass (rxa[channel].bp1.p, 0);
+    xwcpagc (rxa[channel].agc.p);
+    xanf (rxa[channel].anf.p, 1);
+    xanr (rxa[channel].anr.p, 1);
+    xemnr (rxa[channel].emnr.p, 1);
+    xbandpass (rxa[channel].bp1.p, 1);
+    xmeter (rxa[channel].agcmeter.p);
+    xsiphon (rxa[channel].sip1.p, 0);
+    xcbl (rxa[channel].cbl.p);
+    xspeak (rxa[channel].speak.p);
+    xmpeak (rxa[channel].mpeak.p);
+    xssql (rxa[channel].ssql.p);
+    xpanel (rxa[channel].panel.p);
+    xamsq (rxa[channel].amsq.p);
+    xresample (rxa[channel].rsmpout.p);
+}
+
+void setInputSamplerate_rxa (int channel)
+{
+    // buffers
+    _aligned_free (rxa[channel].inbuff);
+    rxa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    // shift
+    setBuffers_shift (rxa[channel].shift.p, rxa[channel].inbuff, rxa[channel].inbuff);
+    setSize_shift (rxa[channel].shift.p, ch[channel].dsp_insize);
+    setSamplerate_shift (rxa[channel].shift.p, ch[channel].in_rate);
+    // input resampler
+    setBuffers_resample (rxa[channel].rsmpin.p, rxa[channel].inbuff, rxa[channel].midbuff);
+    setSize_resample (rxa[channel].rsmpin.p, ch[channel].dsp_insize);
+    setInRate_resample (rxa[channel].rsmpin.p, ch[channel].in_rate);
+    RXAResCheck (channel);
+}
+
+void setOutputSamplerate_rxa (int channel)
+{
+    // buffers
+    _aligned_free (rxa[channel].outbuff);
+    rxa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // output resampler
+    setBuffers_resample (rxa[channel].rsmpout.p, rxa[channel].midbuff, rxa[channel].outbuff);
+    setOutRate_resample (rxa[channel].rsmpout.p, ch[channel].out_rate);
+    RXAResCheck (channel);
+}
+
+void setDSPSamplerate_rxa (int channel)
+{
+    // buffers
+    _aligned_free (rxa[channel].inbuff);
+    rxa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    _aligned_free (rxa[channel].outbuff);
+    rxa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // shift
+    setBuffers_shift (rxa[channel].shift.p, rxa[channel].inbuff, rxa[channel].inbuff);
+    setSize_shift (rxa[channel].shift.p, ch[channel].dsp_insize);
+    // input resampler
+    setBuffers_resample (rxa[channel].rsmpin.p, rxa[channel].inbuff, rxa[channel].midbuff);
+    setSize_resample (rxa[channel].rsmpin.p, ch[channel].dsp_insize);
+    setOutRate_resample (rxa[channel].rsmpin.p, ch[channel].dsp_rate);
+    // dsp_rate blocks
+    setSamplerate_gen (rxa[channel].gen0.p, ch[channel].dsp_rate);
+    setSamplerate_meter (rxa[channel].adcmeter.p, ch[channel].dsp_rate);
+    setSamplerate_nbp (rxa[channel].nbp0.p, ch[channel].dsp_rate);
+    setSamplerate_bpsnba (rxa[channel].bpsnba.p, ch[channel].dsp_rate);
+    setSamplerate_meter (rxa[channel].smeter.p, ch[channel].dsp_rate);
+    setSamplerate_sender (rxa[channel].sender.p, ch[channel].dsp_rate);
+    setSamplerate_amsq (rxa[channel].amsq.p, ch[channel].dsp_rate);
+    setSamplerate_amd (rxa[channel].amd.p, ch[channel].dsp_rate);
+    setSamplerate_fmd (rxa[channel].fmd.p, ch[channel].dsp_rate);
+    setBuffers_fmsq (rxa[channel].fmsq.p, rxa[channel].midbuff, rxa[channel].midbuff, rxa[channel].fmd.p->audio);
+    setSamplerate_fmsq (rxa[channel].fmsq.p, ch[channel].dsp_rate);
+    setSamplerate_snba (rxa[channel].snba.p, ch[channel].dsp_rate);
+    setSamplerate_eqp (rxa[channel].eqp.p, ch[channel].dsp_rate);
+    setSamplerate_anf (rxa[channel].anf.p, ch[channel].dsp_rate);
+    setSamplerate_anr (rxa[channel].anr.p, ch[channel].dsp_rate);
+    setSamplerate_emnr (rxa[channel].emnr.p, ch[channel].dsp_rate);
+    setSamplerate_bandpass (rxa[channel].bp1.p, ch[channel].dsp_rate);
+    setSamplerate_wcpagc (rxa[channel].agc.p, ch[channel].dsp_rate);
+    setSamplerate_meter (rxa[channel].agcmeter.p, ch[channel].dsp_rate);
+    setSamplerate_siphon (rxa[channel].sip1.p, ch[channel].dsp_rate);
+    setSamplerate_cbl (rxa[channel].cbl.p, ch[channel].dsp_rate);
+    setSamplerate_speak (rxa[channel].speak.p, ch[channel].dsp_rate);
+    setSamplerate_mpeak (rxa[channel].mpeak.p, ch[channel].dsp_rate);
+    setSamplerate_ssql (rxa[channel].ssql.p, ch[channel].dsp_rate);
+    setSamplerate_panel (rxa[channel].panel.p, ch[channel].dsp_rate);
+    // output resampler
+    setBuffers_resample (rxa[channel].rsmpout.p, rxa[channel].midbuff, rxa[channel].outbuff);
+    setInRate_resample (rxa[channel].rsmpout.p, ch[channel].dsp_rate);
+    RXAResCheck (channel);
+}
+
+void setDSPBuffsize_rxa (int channel)
+{
+    // buffers
+    _aligned_free(rxa[channel].inbuff);
+    rxa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    _aligned_free (rxa[channel].midbuff);
+    rxa[channel].midbuff = (double *)malloc0(2 * ch[channel].dsp_size * sizeof(complex));
+    _aligned_free (rxa[channel].outbuff);
+    rxa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // shift
+    setBuffers_shift (rxa[channel].shift.p, rxa[channel].inbuff, rxa[channel].inbuff);
+    setSize_shift (rxa[channel].shift.p, ch[channel].dsp_insize);
+    // input resampler
+    setBuffers_resample (rxa[channel].rsmpin.p, rxa[channel].inbuff, rxa[channel].midbuff);
+    setSize_resample (rxa[channel].rsmpin.p, ch[channel].dsp_insize);
+    // dsp_size blocks
+    setBuffers_gen (rxa[channel].gen0.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_gen (rxa[channel].gen0.p, ch[channel].dsp_size);
+    setBuffers_meter (rxa[channel].adcmeter.p, rxa[channel].midbuff);
+    setSize_meter (rxa[channel].adcmeter.p, ch[channel].dsp_size);
+    setBuffers_nbp (rxa[channel].nbp0.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_nbp (rxa[channel].nbp0.p, ch[channel].dsp_size);
+    setBuffers_bpsnba (rxa[channel].bpsnba.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_bpsnba (rxa[channel].bpsnba.p, ch[channel].dsp_size);
+    setBuffers_meter (rxa[channel].smeter.p, rxa[channel].midbuff);
+    setSize_meter (rxa[channel].smeter.p, ch[channel].dsp_size);
+    setBuffers_sender (rxa[channel].sender.p, rxa[channel].midbuff);
+    setSize_sender (rxa[channel].sender.p, ch[channel].dsp_size);
+    setBuffers_amsq (rxa[channel].amsq.p, rxa[channel].midbuff, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_amsq (rxa[channel].amsq.p, ch[channel].dsp_size);
+    setBuffers_amd (rxa[channel].amd.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_amd (rxa[channel].amd.p, ch[channel].dsp_size);
+    setBuffers_fmd (rxa[channel].fmd.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_fmd (rxa[channel].fmd.p, ch[channel].dsp_size);
+    setBuffers_fmsq (rxa[channel].fmsq.p, rxa[channel].midbuff, rxa[channel].midbuff, rxa[channel].fmd.p->audio);
+    setSize_fmsq (rxa[channel].fmsq.p, ch[channel].dsp_size);
+    setBuffers_snba (rxa[channel].snba.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_snba (rxa[channel].snba.p, ch[channel].dsp_size);
+    setBuffers_eqp (rxa[channel].eqp.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_eqp (rxa[channel].eqp.p, ch[channel].dsp_size);
+    setBuffers_anf (rxa[channel].anf.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_anf (rxa[channel].anf.p, ch[channel].dsp_size);
+    setBuffers_anr (rxa[channel].anr.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_anr (rxa[channel].anr.p, ch[channel].dsp_size);
+    setBuffers_emnr (rxa[channel].emnr.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_emnr (rxa[channel].emnr.p, ch[channel].dsp_size);
+    setBuffers_bandpass (rxa[channel].bp1.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_bandpass (rxa[channel].bp1.p, ch[channel].dsp_size);
+    setBuffers_wcpagc (rxa[channel].agc.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_wcpagc (rxa[channel].agc.p, ch[channel].dsp_size);
+    setBuffers_meter (rxa[channel].agcmeter.p, rxa[channel].midbuff);
+    setSize_meter (rxa[channel].agcmeter.p, ch[channel].dsp_size);
+    setBuffers_siphon (rxa[channel].sip1.p, rxa[channel].midbuff);
+    setSize_siphon (rxa[channel].sip1.p, ch[channel].dsp_size);
+    setBuffers_cbl (rxa[channel].cbl.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_cbl (rxa[channel].cbl.p, ch[channel].dsp_size);
+    setBuffers_speak (rxa[channel].speak.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_speak (rxa[channel].speak.p, ch[channel].dsp_size);
+    setBuffers_mpeak (rxa[channel].mpeak.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_mpeak (rxa[channel].mpeak.p, ch[channel].dsp_size);
+    setBuffers_ssql (rxa[channel].ssql.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_ssql (rxa[channel].ssql.p, ch[channel].dsp_size);
+    setBuffers_panel (rxa[channel].panel.p, rxa[channel].midbuff, rxa[channel].midbuff);
+    setSize_panel (rxa[channel].panel.p, ch[channel].dsp_size);
+    // output resampler
+    setBuffers_resample (rxa[channel].rsmpout.p, rxa[channel].midbuff, rxa[channel].outbuff);
+    setSize_resample (rxa[channel].rsmpout.p, ch[channel].dsp_size);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       RXA Mode & Filter Controls                                      *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXAMode (int channel, int mode)
+{
+    if (rxa[channel].mode != mode)
+    {
+        int amd_run = (mode == RXA_AM) || (mode == RXA_SAM);
+        RXAbpsnbaCheck (channel, mode, rxa[channel].ndb.p->master_run);
+        RXAbp1Check (channel, amd_run, rxa[channel].snba.p->run, rxa[channel].emnr.p->run,
+            rxa[channel].anf.p->run, rxa[channel].anr.p->run);
+        EnterCriticalSection (&ch[channel].csDSP);
+        rxa[channel].mode = mode;
+        rxa[channel].amd.p->run  = 0;
+        rxa[channel].fmd.p->run  = 0;
+        rxa[channel].agc.p->run  = 1;
+        switch (mode)
+        {
+        case RXA_AM:
+            rxa[channel].amd.p->run  = 1;
+            rxa[channel].amd.p->mode = 0;
+            break;
+        case RXA_SAM:
+            rxa[channel].amd.p->run  = 1;
+            rxa[channel].amd.p->mode = 1;
+            break;
+        case RXA_DSB:
+
+            break;
+        case RXA_FM:
+            rxa[channel].fmd.p->run  = 1;
+            rxa[channel].agc.p->run  = 0;
+            break;
+        default:
+
+            break;
+        }
+        RXAbp1Set (channel);
+        RXAbpsnbaSet (channel);                         // update variables
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+void RXAResCheck (int channel)
+{
+    // turn OFF/ON resamplers depending upon whether they're needed
+    RESAMPLE a = rxa[channel].rsmpin.p;
+    if (ch[channel].in_rate  != ch[channel].dsp_rate)   a->run = 1;
+    else                                                a->run = 0;
+    a = rxa[channel].rsmpout.p;
+    if (ch[channel].dsp_rate != ch[channel].out_rate)   a->run = 1;
+    else                                                a->run = 0;
+}
+
+void RXAbp1Check (int channel, int amd_run, int snba_run,
+    int emnr_run, int anf_run, int anr_run)
+{
+    BANDPASS a = rxa[channel].bp1.p;
+    double gain;
+    if (amd_run  ||
+        snba_run ||
+        emnr_run ||
+        anf_run  ||
+        anr_run)    gain = 2.0;
+    else            gain = 1.0;
+    if (a->gain != gain)
+        setGain_bandpass (a, gain, 0);
+}
+
+void RXAbp1Set (int channel)
+{
+    BANDPASS a = rxa[channel].bp1.p;
+    int old = a->run;
+    if ((rxa[channel].amd.p->run  == 1) ||
+        (rxa[channel].snba.p->run == 1) ||
+        (rxa[channel].emnr.p->run == 1) ||
+        (rxa[channel].anf.p->run  == 1) ||
+        (rxa[channel].anr.p->run  == 1))    a->run = 1;
+    else                                    a->run = 0;
+    if (!old && a->run) flush_bandpass (a);
+    setUpdate_fircore (a->p);
+}
+
+void RXAbpsnbaCheck (int channel, int mode, int notch_run)
+{
+    // for BPSNBA: set run, position, freqs, run_notches
+    // call this upon change in RXA_mode, snba_run, notch_master_run
+    BPSNBA a = rxa[channel].bpsnba.p;
+    double f_low = 0.0, f_high = 0.0;
+    int run_notches = 0;
+    switch (mode)
+    {
+        case RXA_LSB:
+        case RXA_CWL:
+        case RXA_DIGL:
+            f_low  = -a->abs_high_freq;
+            f_high = -a->abs_low_freq;
+            run_notches = notch_run;
+            break;
+        case RXA_USB:
+        case RXA_CWU:
+        case RXA_DIGU:
+            f_low  = +a->abs_low_freq;
+            f_high = +a->abs_high_freq;
+            run_notches = notch_run;
+            break;
+        case RXA_AM:
+        case RXA_SAM:
+        case RXA_DSB:
+            f_low  = +a->abs_low_freq;
+            f_high = +a->abs_high_freq;
+            run_notches = 0;
+            break;
+        case RXA_FM:
+            f_low  = +a->abs_low_freq;
+            f_high = +a->abs_high_freq;
+            run_notches = 0;
+            break;
+        case RXA_DRM:
+        case RXA_SPEC:
+
+            break;
+    }
+    // 'run' and 'position' are examined at run time; no filter changes required.
+    // Recalculate filter if frequencies OR 'run_notches' changed.
+    if ((a->f_low       != f_low      ) ||
+        (a->f_high      != f_high     ) ||
+        (a->run_notches != run_notches))
+    {
+        a->f_low  = f_low;
+        a->f_high = f_high;
+        a->run_notches = run_notches;
+        // f_low, f_high, run_notches are needed for the filter recalculation
+        recalc_bpsnba_filter (a, 0);
+    }
+}
+
+void RXAbpsnbaSet (int channel)
+{
+    // for BPSNBA: set run, position, freqs, run_notches
+    // call this upon change in RXA_mode, snba_run, notch_master_run
+    BPSNBA a = rxa[channel].bpsnba.p;
+    switch (rxa[channel].mode)
+    {
+        case RXA_LSB:
+        case RXA_CWL:
+        case RXA_DIGL:
+            a->run = rxa[channel].snba.p->run;
+            a->position = 0;
+            break;
+        case RXA_USB:
+        case RXA_CWU:
+        case RXA_DIGU:
+            a->run = rxa[channel].snba.p->run;
+            a->position = 0;
+            break;
+        case RXA_AM:
+        case RXA_SAM:
+        case RXA_DSB:
+            a->run = rxa[channel].snba.p->run;
+            a->position = 1;
+            break;
+        case RXA_FM:
+            a->run = rxa[channel].snba.p->run;
+            a->position = 1;
+            break;
+        case RXA_DRM:
+        case RXA_SPEC:
+            a->run = 0;
+            break;
+    }
+    setUpdate_fircore (a->bpsnba->p);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                               Collectives                                             *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void RXASetPassband (int channel, double f_low, double f_high)
+{
+    SetRXABandpassFreqs         (channel, f_low, f_high);
+    SetRXASNBAOutputBandwidth   (channel, f_low, f_high);
+    RXANBPSetFreqs              (channel, f_low, f_high);
+}
+
+PORT
+void RXASetNC (int channel, int nc)
+{
+    int oldstate = SetChannelState (channel, 0, 1);
+    RXANBPSetNC                 (channel, nc);
+    RXABPSNBASetNC              (channel, nc);
+    SetRXABandpassNC            (channel, nc);
+    SetRXAEQNC                  (channel, nc);
+    SetRXAFMSQNC                (channel, nc);
+    SetRXAFMNCde                (channel, nc);
+    SetRXAFMNCaud               (channel, nc);
+    SetChannelState (channel, oldstate, 0);
+}
+
+PORT
+void RXASetMP (int channel, int mp)
+{
+    RXANBPSetMP                 (channel, mp);
+    RXABPSNBASetMP              (channel, mp);
+    SetRXABandpassMP            (channel, mp);
+    SetRXAEQMP                  (channel, mp);
+    SetRXAFMSQMP                (channel, mp);
+    SetRXAFMMPde                (channel, mp);
+    SetRXAFMMPaud               (channel, mp);
+}

wdsp/RXA.h

@@ -0,0 +1,201 @@
+/*  RXA.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2014, 2015, 2016 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#ifndef _rxa_h
+#define _rxa_h
+#include "comm.h"
+
+enum rxaMode
+{
+    RXA_LSB,
+    RXA_USB,
+    RXA_DSB,
+    RXA_CWL,
+    RXA_CWU,
+    RXA_FM,
+    RXA_AM,
+    RXA_DIGU,
+    RXA_SPEC,
+    RXA_DIGL,
+    RXA_SAM,
+    RXA_DRM
+};
+
+enum rxaMeterType
+{
+    RXA_S_PK,
+    RXA_S_AV,
+    RXA_ADC_PK,
+    RXA_ADC_AV,
+    RXA_AGC_GAIN,
+    RXA_AGC_PK,
+    RXA_AGC_AV,
+    RXA_METERTYPE_LAST
+};
+
+struct _rxa
+{
+    double* inbuff;
+    double* outbuff;
+    double* midbuff;
+    int mode;
+    double meter[RXA_METERTYPE_LAST];
+    CRITICAL_SECTION* pmtupdate[RXA_METERTYPE_LAST];
+    struct
+    {
+        METER p;
+    } smeter, adcmeter, agcmeter;
+    struct
+    {
+        SHIFT p;
+    } shift;
+    struct
+    {
+        RESAMPLE p;
+    } rsmpin, rsmpout;
+    struct
+    {
+        GEN p;
+    } gen0;
+    struct
+    {
+        BANDPASS p;
+    } bp1;
+    struct
+    {
+        NOTCHDB p;
+    } ndb;
+    struct
+    {
+        NBP p;
+    } nbp0;
+    struct
+    {
+        BPSNBA p;
+    } bpsnba;
+    struct
+    {
+        SNBA p;
+    } snba;
+    struct
+    {
+        SENDER p;
+    } sender;
+    struct
+    {
+        AMSQ p;
+    } amsq;
+    struct
+    {
+        AMD p;
+    } amd;
+    struct
+    {
+        FMD p;
+    } fmd;
+    struct
+    {
+        FMSQ p;
+    } fmsq;
+    struct
+    {
+        EQP p;
+    } eqp;
+    struct
+    {
+        ANF p;
+    } anf;
+    struct
+    {
+        ANR p;
+    } anr;
+    struct
+    {
+        EMNR p;
+    } emnr;
+    struct
+    {
+        WCPAGC p;
+    } agc;
+    struct
+    {
+        SPEAK p;
+    } speak;
+    struct
+    {
+        MPEAK p;
+    } mpeak;
+    struct
+    {
+        PANEL p;
+    } panel;
+    struct
+    {
+        SIPHON p;
+    } sip1;
+    struct
+    {
+        CBL p;
+    } cbl;
+    struct
+    {
+        SSQL p;
+    } ssql;
+};
+
+extern struct _rxa rxa[];
+
+extern void create_rxa (int channel);
+
+extern void destroy_rxa (int channel);
+
+extern void flush_rxa (int channel);
+
+extern void xrxa (int channel);
+
+extern void setInputSamplerate_rxa (int channel);
+
+extern void setOutputSamplerate_rxa (int channel);
+
+extern void setDSPSamplerate_rxa (int channel);
+
+extern void setDSPBuffsize_rxa (int channel);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAMode (int channel, int mode);
+
+extern void RXAResCheck (int channel);
+
+extern void RXAbp1Check (int channel, int amd_run, int snba_run, int emnr_run, int anf_run, int anr_run);
+
+extern void RXAbp1Set (int channel);
+
+extern void RXAbpsnbaCheck (int channel, int mode, int notch_run);
+
+extern void RXAbpsnbaSet (int channel);
+
+#endif

wdsp/TXA.c

@@ -0,0 +1,935 @@
+/*  TXA.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2014, 2016, 2017, 2021, 2023 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#include "comm.h"
+
+struct _txa txa[MAX_CHANNELS];
+
+void create_txa (int channel)
+{
+    txa[channel].mode   = TXA_LSB;
+    txa[channel].f_low  = -5000.0;
+    txa[channel].f_high = - 100.0;
+    txa[channel].inbuff  = (double *) malloc0 (1 * ch[channel].dsp_insize  * sizeof (complex));
+    txa[channel].outbuff = (double *) malloc0 (1 * ch[channel].dsp_outsize * sizeof (complex));
+    txa[channel].midbuff = (double *) malloc0 (2 * ch[channel].dsp_size    * sizeof (complex));
+
+    txa[channel].rsmpin.p = create_resample (
+        0,                                          // run - will be turned on below if needed
+        ch[channel].dsp_insize,                     // input buffer size
+        txa[channel].inbuff,                        // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        ch[channel].in_rate,                        // input sample rate
+        ch[channel].dsp_rate,                       // output sample rate
+        0.0,                                        // select cutoff automatically
+        0,                                          // select ncoef automatically
+        1.0);                                       // gain
+
+    txa[channel].gen0.p = create_gen (
+        0,                                          // run
+        ch[channel].dsp_size,                       // buffer size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // sample rate
+        2);                                         // mode
+
+    txa[channel].panel.p = create_panel (
+        channel,                                    // channel number
+        1,                                          // run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        1.0,                                        // gain1
+        1.0,                                        // gain2I
+        1.0,                                        // gain2Q
+        2,                                          // 1 to use Q, 2 to use I for input
+        0);                                         // 0, no copy
+
+    txa[channel].phrot.p = create_phrot (
+        0,                                          // run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // samplerate
+        338.0,                                      // 1/2 of phase frequency
+        8);                                         // number of stages
+
+    txa[channel].micmeter.p = create_meter (
+        1,                                          // run
+        0,                                          // optional pointer to another 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_MIC_AV,                                 // index for average value
+        TXA_MIC_PK,                                 // index for peak value
+        -1,                                         // index for gain value
+        0);                                         // pointer for gain computation
+
+    txa[channel].amsq.p = create_amsq (
+        0,                                          // run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        txa[channel].midbuff,                       // trigger buffer
+        ch[channel].dsp_rate,                       // sample rate
+        0.010,                                      // time constant for averaging signal
+        0.004,                                      // up-slew time
+        0.004,                                      // down-slew time
+        0.180,                                      // signal level to initiate tail
+        0.200,                                      // signal level to initiate unmute
+        0.000,                                      // minimum tail length
+        0.025,                                      // maximum tail length
+        0.200);                                     // muted gain
+
+    {
+    double default_F[11] = {0.0,  32.0,  63.0, 125.0, 250.0, 500.0, 1000.0, 2000.0, 4000.0, 8000.0, 16000.0};
+    double default_G[11] = {0.0, -12.0, -12.0, -12.0,  -1.0,  +1.0,   +4.0,   +9.0,  +12.0,  -10.0,   -10.0};
+    //double default_G[11] =   {0.0,   0.0,   0.0,   0.0,   0.0,   0.0,    0.0,    0.0,    0.0,    0.0,     0.0};
+    txa[channel].eqp.p = create_eqp (
+        0,                                          // run - OFF by default
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of filter coefficients
+        0,                                          // minimum phase flag
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        10,                                         // nfreqs
+        default_F,                                  // vector of frequencies
+        default_G,                                  // vector of gain values
+        0,                                          // cutoff mode
+        0,                                          // wintype
+        ch[channel].dsp_rate);                      // samplerate
+    }
+
+    txa[channel].eqmeter.p = create_meter (
+        1,                                          // run
+        &(txa[channel].eqp.p->run),                 // pointer to eqp 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_EQ_AV,                                  // index for average value
+        TXA_EQ_PK,                                  // index for peak value
+        -1,                                         // index for gain value
+        0);                                         // pointer for gain computation
+
+    txa[channel].preemph.p = create_emphp (
+        0,                                          // run
+        1,                                          // position
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of filter coefficients
+        0,                                          // minimum phase flag
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer,
+        ch[channel].dsp_rate,                       // sample rate
+        0,                                          // pre-emphasis type
+        300.0,                                      // f_low
+        3000.0);                                    // f_high
+
+    txa[channel].leveler.p = create_wcpagc (
+        0,                                          // run - OFF by default
+        5,                                          // mode
+        0,                                          // 0 for max(I,Q), 1 for envelope
+        txa[channel].midbuff,                       // input buff pointer
+        txa[channel].midbuff,                       // output buff pointer
+        ch[channel].dsp_size,                       // io_buffsize
+        ch[channel].dsp_rate,                       // sample rate
+        0.001,                                      // tau_attack
+        0.500,                                      // tau_decay
+        6,                                          // n_tau
+        1.778,                                      // max_gain
+        1.0,                                        // var_gain
+        1.0,                                        // fixed_gain
+        1.0,                                        // max_input
+        1.05,                                       // out_targ
+        0.250,                                      // tau_fast_backaverage
+        0.005,                                      // tau_fast_decay
+        5.0,                                        // pop_ratio
+        0,                                          // hang_enable
+        0.500,                                      // tau_hang_backmult
+        0.500,                                      // hangtime
+        2.000,                                      // hang_thresh
+        0.100);                                     // tau_hang_decay
+
+    txa[channel].lvlrmeter.p = create_meter (
+        1,                                          // run
+        &(txa[channel].leveler.p->run),             // pointer to leveler 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_LVLR_AV,                                // index for average value
+        TXA_LVLR_PK,                                // index for peak value
+        TXA_LVLR_GAIN,                              // index for gain value
+        &txa[channel].leveler.p->gain);             // pointer for gain computation
+
+    {
+    double default_F[5] = {200.0, 1000.0, 2000.0, 3000.0, 4000.0};
+    double default_G[5] = {0.0, 5.0, 10.0, 10.0, 5.0};
+    double default_E[5] = {7.0, 7.0, 7.0, 7.0, 7.0};
+    txa[channel].cfcomp.p = create_cfcomp(
+        0,                                          // run
+        0,                                          // position
+        0,                                          // post-equalizer run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        2048,                                       // fft size
+        4,                                          // overlap
+        ch[channel].dsp_rate,                       // samplerate
+        1,                                          // window type
+        0,                                          // compression method
+        5,                                          // nfreqs
+        0.0,                                        // pre-compression
+        0.0,                                        // pre-postequalization
+        default_F,                                  // frequency array
+        default_G,                                  // compression array
+        default_E,                                  // eq array
+        0.25,                                       // metering time constant
+        0.50);                                      // display time constant
+    }
+
+    txa[channel].cfcmeter.p = create_meter (
+        1,                                          // run
+        &(txa[channel].cfcomp.p->run),              // pointer to eqp 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_CFC_AV,                                 // index for average value
+        TXA_CFC_PK,                                 // index for peak value
+        TXA_CFC_GAIN,                               // index for gain value
+        &txa[channel].cfcomp.p->gain);              // pointer for gain computation
+
+    txa[channel].bp0.p = create_bandpass (
+        1,                                          // always runs
+        0,                                          // position
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of coefficients
+        0,                                          // flag for minimum phase
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        txa[channel].f_low,                         // low freq cutoff
+        txa[channel].f_high,                        // high freq cutoff
+        ch[channel].dsp_rate,                       // samplerate
+        1,                                          // wintype
+        2.0);                                       // gain
+
+    txa[channel].compressor.p = create_compressor (
+        0,                                          // run - OFF by default
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        3.0);                                       // gain
+
+    txa[channel].bp1.p = create_bandpass (
+        0,                                          // ONLY RUNS WHEN COMPRESSOR IS USED
+        0,                                          // position
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of coefficients
+        0,                                          // flag for minimum phase
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        txa[channel].f_low,                         // low freq cutoff
+        txa[channel].f_high,                        // high freq cutoff
+        ch[channel].dsp_rate,                       // samplerate
+        1,                                          // wintype
+        2.0);                                       // gain
+
+    txa[channel].osctrl.p = create_osctrl (
+        0,                                          // run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // sample rate
+        1.95);                                      // gain for clippings
+
+    txa[channel].bp2.p = create_bandpass (
+        0,                                          // ONLY RUNS WHEN COMPRESSOR IS USED
+        0,                                          // position
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of coefficients
+        0,                                          // flag for minimum phase
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        txa[channel].f_low,                         // low freq cutoff
+        txa[channel].f_high,                        // high freq cutoff
+        ch[channel].dsp_rate,                       // samplerate
+        1,                                          // wintype
+        1.0);                                       // gain
+
+    txa[channel].compmeter.p = create_meter (
+        1,                                          // run
+        &(txa[channel].compressor.p->run),          // pointer to compressor 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_COMP_AV,                                // index for average value
+        TXA_COMP_PK,                                // index for peak value
+        -1,                                         // index for gain value
+        0);                                         // pointer for gain computation
+
+    txa[channel].alc.p = create_wcpagc (
+        1,                                          // run - always ON
+        5,                                          // mode
+        1,                                          // 0 for max(I,Q), 1 for envelope
+        txa[channel].midbuff,                       // input buff pointer
+        txa[channel].midbuff,                       // output buff pointer
+        ch[channel].dsp_size,                       // io_buffsize
+        ch[channel].dsp_rate,                       // sample rate
+        0.001,                                      // tau_attack
+        0.010,                                      // tau_decay
+        6,                                          // n_tau
+        1.0,                                        // max_gain
+        1.0,                                        // var_gain
+        1.0,                                        // fixed_gain
+        1.0,                                        // max_input
+        1.0,                                        // out_targ
+        0.250,                                      // tau_fast_backaverage
+        0.005,                                      // tau_fast_decay
+        5.0,                                        // pop_ratio
+        0,                                          // hang_enable
+        0.500,                                      // tau_hang_backmult
+        0.500,                                      // hangtime
+        2.000,                                      // hang_thresh
+        0.100);                                     // tau_hang_decay
+
+    txa[channel].ammod.p = create_ammod (
+        0,                                          // run - OFF by default
+        0,                                          // mode:  0=>AM, 1=>DSB
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to output buffer
+        0.5);                                       // carrier level
+
+
+    txa[channel].fmmod.p = create_fmmod (
+        0,                                          // run - OFF by default
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].midbuff,                       // pointer to input buffer
+        ch[channel].dsp_rate,                       // samplerate
+        5000.0,                                     // deviation
+        300.0,                                      // low cutoff frequency
+        3000.0,                                     // high cutoff frequency
+        1,                                          // ctcss run control
+        0.10,                                       // ctcss level
+        100.0,                                      // ctcss frequency
+        1,                                          // run bandpass filter
+        max(2048, ch[channel].dsp_size),            // number coefficients for bandpass filter
+        0);                                         // minimum phase flag
+
+    txa[channel].gen1.p = create_gen (
+        0,                                          // run
+        ch[channel].dsp_size,                       // buffer size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // sample rate
+        0);                                         // mode
+
+    txa[channel].uslew.p = create_uslew (
+        channel,                                    // channel
+        &ch[channel].iob.ch_upslew,                 // pointer to channel upslew flag
+        ch[channel].dsp_size,                       // buffer size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // sample rate
+        0.000,                                      // delay time
+        0.005);                                     // upslew time
+
+        txa[channel].alcmeter.p = create_meter (
+        1,                                          // run
+        0,                                          // optional pointer to a 'run'
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // pointer to buffer
+        ch[channel].dsp_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_ALC_AV,                                 // index for average value
+        TXA_ALC_PK,                                 // index for peak value
+        TXA_ALC_GAIN,                               // index for gain value
+        &txa[channel].alc.p->gain);                 // pointer for gain computation
+
+    txa[channel].sip1.p = create_siphon (
+        1,                                          // run
+        0,                                          // position
+        0,                                          // mode
+        0,                                          // disp
+        ch[channel].dsp_size,                       // input buffer size
+        txa[channel].midbuff,                       // input buffer
+        16384,                                      // number of samples to buffer
+        16384,                                      // fft size for spectrum
+        1);                                         // specmode
+
+    txa[channel].calcc.p = create_calcc (
+        channel,                                    // channel number
+        1,                                          // run calibration
+        1024,                                       // input buffer size
+        ch[channel].in_rate,                        // samplerate
+        16,                                         // ints
+        256,                                        // spi
+        (1.0 / 0.4072),                             // hw_scale
+        0.1,                                        // mox delay
+        0.0,                                        // loop delay
+        0.8,                                        // ptol
+        0,                                          // mox
+        0,                                          // solidmox
+        1,                                          // pin mode
+        1,                                          // map mode
+        0,                                          // stbl mode
+        256,                                        // pin samples
+        0.9);                                       // alpha
+
+    txa[channel].iqc.p0 = txa[channel].iqc.p1 = create_iqc (
+        0,                                          // run
+        ch[channel].dsp_size,                       // size
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        (double)ch[channel].dsp_rate,               // sample rate
+        16,                                         // ints
+        0.005,                                      // changeover time
+        256);                                       // spi
+
+    txa[channel].cfir.p = create_cfir(
+        0,                                          // run
+        ch[channel].dsp_size,                       // size
+        max(2048, ch[channel].dsp_size),            // number of filter coefficients
+        0,                                          // minimum phase flag
+        txa[channel].midbuff,                       // input buffer
+        txa[channel].midbuff,                       // output buffer
+        ch[channel].dsp_rate,                       // input sample rate
+        ch[channel].out_rate,                       // CIC input sample rate
+        1,                                          // CIC differential delay
+        640,                                        // CIC interpolation factor
+        5,                                          // CIC integrator-comb pairs
+        20000.0,                                    // cutoff frequency
+        2,                                          // brick-wall windowed rolloff
+        0.0,                                        // raised-cosine transition width
+        0);                                         // window type
+
+    txa[channel].rsmpout.p = create_resample (
+        0,                                          // run - will be turned ON below if needed
+        ch[channel].dsp_size,                       // input size
+        txa[channel].midbuff,                       // pointer to input buffer
+        txa[channel].outbuff,                       // pointer to output buffer
+        ch[channel].dsp_rate,                       // input sample rate
+        ch[channel].out_rate,                       // output sample rate
+        0.0,                                        // select cutoff automatically
+        0,                                          // select ncoef automatically
+        0.980);                                     // gain
+
+    txa[channel].outmeter.p = create_meter (
+        1,                                          // run
+        0,                                          // optional pointer to another 'run'
+        ch[channel].dsp_outsize,                    // size
+        txa[channel].outbuff,                       // pointer to buffer
+        ch[channel].out_rate,                       // samplerate
+        0.100,                                      // averaging time constant
+        0.100,                                      // peak decay time constant
+        txa[channel].meter,                         // result vector
+        txa[channel].pmtupdate,                     // locks for meter access
+        TXA_OUT_AV,                                 // index for average value
+        TXA_OUT_PK,                                 // index for peak value
+        -1,                                         // index for gain value
+        0);                                         // pointer for gain computation
+
+    // turn OFF / ON resamplers as needed
+    TXAResCheck (channel);
+}
+
+void destroy_txa (int channel)
+{
+    // in reverse order, free each item we created
+    destroy_meter (txa[channel].outmeter.p);
+    destroy_resample (txa[channel].rsmpout.p);
+    destroy_cfir(txa[channel].cfir.p);
+    destroy_calcc (txa[channel].calcc.p);
+    destroy_iqc (txa[channel].iqc.p0);
+    destroy_siphon (txa[channel].sip1.p);
+    destroy_meter (txa[channel].alcmeter.p);
+    destroy_uslew (txa[channel].uslew.p);
+    destroy_gen (txa[channel].gen1.p);
+    destroy_fmmod (txa[channel].fmmod.p);
+    destroy_ammod (txa[channel].ammod.p);
+    destroy_wcpagc (txa[channel].alc.p);
+    destroy_meter (txa[channel].compmeter.p);
+    destroy_bandpass (txa[channel].bp2.p);
+    destroy_osctrl (txa[channel].osctrl.p);
+    destroy_bandpass (txa[channel].bp1.p);
+    destroy_compressor (txa[channel].compressor.p);
+    destroy_bandpass (txa[channel].bp0.p);
+    destroy_meter (txa[channel].cfcmeter.p);
+    destroy_cfcomp (txa[channel].cfcomp.p);
+    destroy_meter (txa[channel].lvlrmeter.p);
+    destroy_wcpagc (txa[channel].leveler.p);
+    destroy_emphp (txa[channel].preemph.p);
+    destroy_meter (txa[channel].eqmeter.p);
+    destroy_eqp (txa[channel].eqp.p);
+    destroy_amsq (txa[channel].amsq.p);
+    destroy_meter (txa[channel].micmeter.p);
+    destroy_phrot (txa[channel].phrot.p);
+    destroy_panel (txa[channel].panel.p);
+    destroy_gen (txa[channel].gen0.p);
+    destroy_resample (txa[channel].rsmpin.p);
+    _aligned_free (txa[channel].midbuff);
+    _aligned_free (txa[channel].outbuff);
+    _aligned_free (txa[channel].inbuff);
+}
+
+void flush_txa (int channel)
+{
+    memset (txa[channel].inbuff,  0, 1 * ch[channel].dsp_insize  * sizeof (complex));
+    memset (txa[channel].outbuff, 0, 1 * ch[channel].dsp_outsize * sizeof (complex));
+    memset (txa[channel].midbuff, 0, 2 * ch[channel].dsp_size    * sizeof (complex));
+    flush_resample (txa[channel].rsmpin.p);
+    flush_gen (txa[channel].gen0.p);
+    flush_panel (txa[channel].panel.p);
+    flush_phrot (txa[channel].phrot.p);
+    flush_meter (txa[channel].micmeter.p);
+    flush_amsq (txa[channel].amsq.p);
+    flush_eqp (txa[channel].eqp.p);
+    flush_meter (txa[channel].eqmeter.p);
+    flush_emphp (txa[channel].preemph.p);
+    flush_wcpagc (txa[channel].leveler.p);
+    flush_meter (txa[channel].lvlrmeter.p);
+    flush_cfcomp (txa[channel].cfcomp.p);
+    flush_meter (txa[channel].cfcmeter.p);
+    flush_bandpass (txa[channel].bp0.p);
+    flush_compressor (txa[channel].compressor.p);
+    flush_bandpass (txa[channel].bp1.p);
+    flush_osctrl (txa[channel].osctrl.p);
+    flush_bandpass (txa[channel].bp2.p);
+    flush_meter (txa[channel].compmeter.p);
+    flush_wcpagc (txa[channel].alc.p);
+    flush_ammod (txa[channel].ammod.p);
+    flush_fmmod (txa[channel].fmmod.p);
+    flush_gen (txa[channel].gen1.p);
+    flush_uslew (txa[channel].uslew.p);
+    flush_meter (txa[channel].alcmeter.p);
+    flush_siphon (txa[channel].sip1.p);
+    flush_iqc (txa[channel].iqc.p0);
+    flush_cfir(txa[channel].cfir.p);
+    flush_resample (txa[channel].rsmpout.p);
+    flush_meter (txa[channel].outmeter.p);
+}
+
+void xtxa (int channel)
+{
+    xresample (txa[channel].rsmpin.p);              // input resampler
+    xgen (txa[channel].gen0.p);                     // input signal generator
+    xpanel (txa[channel].panel.p);                  // includes MIC gain
+    xphrot (txa[channel].phrot.p);                  // phase rotator
+    xmeter (txa[channel].micmeter.p);               // MIC meter
+    xamsqcap (txa[channel].amsq.p);                 // downward expander capture
+    xamsq (txa[channel].amsq.p);                    // downward expander action
+    xeqp (txa[channel].eqp.p);                      // pre-EQ
+    xmeter (txa[channel].eqmeter.p);                // EQ meter
+    xemphp (txa[channel].preemph.p, 0);             // FM pre-emphasis (first option)
+    xwcpagc (txa[channel].leveler.p);               // Leveler
+    xmeter (txa[channel].lvlrmeter.p);              // Leveler Meter
+    xcfcomp (txa[channel].cfcomp.p, 0);             // Continuous Frequency Compressor with post-EQ
+    xmeter (txa[channel].cfcmeter.p);               // CFC+PostEQ Meter
+    xbandpass (txa[channel].bp0.p, 0);              // primary bandpass filter
+    xcompressor (txa[channel].compressor.p);        // COMP compressor
+    xbandpass (txa[channel].bp1.p, 0);              // aux bandpass (runs if COMP)
+    xosctrl (txa[channel].osctrl.p);                // CESSB Overshoot Control
+    xbandpass (txa[channel].bp2.p, 0);              // aux bandpass (runs if CESSB)
+    xmeter (txa[channel].compmeter.p);              // COMP meter
+    xwcpagc (txa[channel].alc.p);                   // ALC
+    xammod (txa[channel].ammod.p);                  // AM Modulator
+    xemphp (txa[channel].preemph.p, 1);             // FM pre-emphasis (second option)
+    xfmmod (txa[channel].fmmod.p);                  // FM Modulator
+    xgen (txa[channel].gen1.p);                     // output signal generator (TUN and Two-tone)
+    xuslew (txa[channel].uslew.p);                  // up-slew for AM, FM, and gens
+    xmeter (txa[channel].alcmeter.p);               // ALC Meter
+    xsiphon (txa[channel].sip1.p, 0);               // siphon data for display
+    xiqc (txa[channel].iqc.p0);                     // PureSignal correction
+    xcfir(txa[channel].cfir.p);                     // compensating FIR filter (used Protocol_2 only)
+    xresample (txa[channel].rsmpout.p);             // output resampler
+    xmeter (txa[channel].outmeter.p);               // output meter
+    // print_peak_env ("env_exception.txt", ch[channel].dsp_outsize, txa[channel].outbuff, 0.7);
+}
+
+void setInputSamplerate_txa (int channel)
+{
+    // buffers
+    _aligned_free (txa[channel].inbuff);
+    txa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    // input resampler
+    setBuffers_resample (txa[channel].rsmpin.p, txa[channel].inbuff, txa[channel].midbuff);
+    setSize_resample (txa[channel].rsmpin.p, ch[channel].dsp_insize);
+    setInRate_resample (txa[channel].rsmpin.p, ch[channel].in_rate);
+    TXAResCheck (channel);
+}
+
+void setOutputSamplerate_txa (int channel)
+{
+    // buffers
+    _aligned_free (txa[channel].outbuff);
+    txa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // cfir - needs to know input rate of firmware CIC
+    setOutRate_cfir (txa[channel].cfir.p, ch[channel].out_rate);
+    // output resampler
+    setBuffers_resample (txa[channel].rsmpout.p, txa[channel].midbuff, txa[channel].outbuff);
+    setOutRate_resample (txa[channel].rsmpout.p, ch[channel].out_rate);
+    TXAResCheck (channel);
+    // output meter
+    setBuffers_meter (txa[channel].outmeter.p, txa[channel].outbuff);
+    setSize_meter (txa[channel].outmeter.p, ch[channel].dsp_outsize);
+    setSamplerate_meter (txa[channel].outmeter.p, ch[channel].out_rate);
+}
+
+void setDSPSamplerate_txa (int channel)
+{
+    // buffers
+    _aligned_free (txa[channel].inbuff);
+    txa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    _aligned_free (txa[channel].outbuff);
+    txa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // input resampler
+    setBuffers_resample (txa[channel].rsmpin.p, txa[channel].inbuff, txa[channel].midbuff);
+    setSize_resample (txa[channel].rsmpin.p, ch[channel].dsp_insize);
+    setOutRate_resample (txa[channel].rsmpin.p, ch[channel].dsp_rate);
+    // dsp_rate blocks
+    setSamplerate_gen (txa[channel].gen0.p, ch[channel].dsp_rate);
+    setSamplerate_panel (txa[channel].panel.p, ch[channel].dsp_rate);
+    setSamplerate_phrot (txa[channel].phrot.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].micmeter.p, ch[channel].dsp_rate);
+    setSamplerate_amsq (txa[channel].amsq.p, ch[channel].dsp_rate);
+    setSamplerate_eqp (txa[channel].eqp.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].eqmeter.p, ch[channel].dsp_rate);
+    setSamplerate_emphp (txa[channel].preemph.p, ch[channel].dsp_rate);
+    setSamplerate_wcpagc (txa[channel].leveler.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].lvlrmeter.p, ch[channel].dsp_rate);
+    setSamplerate_cfcomp (txa[channel].cfcomp.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].cfcmeter.p, ch[channel].dsp_rate);
+    setSamplerate_bandpass (txa[channel].bp0.p, ch[channel].dsp_rate);
+    setSamplerate_compressor (txa[channel].compressor.p, ch[channel].dsp_rate);
+    setSamplerate_bandpass (txa[channel].bp1.p, ch[channel].dsp_rate);
+    setSamplerate_osctrl (txa[channel].osctrl.p, ch[channel].dsp_rate);
+    setSamplerate_bandpass (txa[channel].bp2.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].compmeter.p, ch[channel].dsp_rate);
+    setSamplerate_wcpagc (txa[channel].alc.p, ch[channel].dsp_rate);
+    setSamplerate_ammod (txa[channel].ammod.p, ch[channel].dsp_rate);
+    setSamplerate_fmmod (txa[channel].fmmod.p, ch[channel].dsp_rate);
+    setSamplerate_gen (txa[channel].gen1.p, ch[channel].dsp_rate);
+    setSamplerate_uslew (txa[channel].uslew.p, ch[channel].dsp_rate);
+    setSamplerate_meter (txa[channel].alcmeter.p, ch[channel].dsp_rate);
+    setSamplerate_siphon (txa[channel].sip1.p, ch[channel].dsp_rate);
+    setSamplerate_iqc (txa[channel].iqc.p0, ch[channel].dsp_rate);
+    setSamplerate_cfir (txa[channel].cfir.p, ch[channel].dsp_rate);
+    // output resampler
+    setBuffers_resample (txa[channel].rsmpout.p, txa[channel].midbuff, txa[channel].outbuff);
+    setInRate_resample (txa[channel].rsmpout.p, ch[channel].dsp_rate);
+    TXAResCheck (channel);
+    // output meter
+    setBuffers_meter (txa[channel].outmeter.p, txa[channel].outbuff);
+    setSize_meter (txa[channel].outmeter.p, ch[channel].dsp_outsize);
+}
+
+void setDSPBuffsize_txa (int channel)
+{
+    // buffers
+    _aligned_free (txa[channel].inbuff);
+    txa[channel].inbuff = (double *)malloc0(1 * ch[channel].dsp_insize  * sizeof(complex));
+    _aligned_free (txa[channel].midbuff);
+    txa[channel].midbuff = (double *)malloc0(2 * ch[channel].dsp_size * sizeof(complex));
+    _aligned_free (txa[channel].outbuff);
+    txa[channel].outbuff = (double *)malloc0(1 * ch[channel].dsp_outsize * sizeof(complex));
+    // input resampler
+    setBuffers_resample (txa[channel].rsmpin.p, txa[channel].inbuff, txa[channel].midbuff);
+    setSize_resample (txa[channel].rsmpin.p, ch[channel].dsp_insize);
+    // dsp_size blocks
+    setBuffers_gen (txa[channel].gen0.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_gen (txa[channel].gen0.p, ch[channel].dsp_size);
+    setBuffers_panel (txa[channel].panel.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_panel (txa[channel].panel.p, ch[channel].dsp_size);
+    setBuffers_phrot (txa[channel].phrot.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_phrot (txa[channel].phrot.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].micmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].micmeter.p, ch[channel].dsp_size);
+    setBuffers_amsq (txa[channel].amsq.p, txa[channel].midbuff, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_amsq (txa[channel].amsq.p, ch[channel].dsp_size);
+    setBuffers_eqp (txa[channel].eqp.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_eqp (txa[channel].eqp.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].eqmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].eqmeter.p, ch[channel].dsp_size);
+    setBuffers_emphp (txa[channel].preemph.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_emphp (txa[channel].preemph.p, ch[channel].dsp_size);
+    setBuffers_wcpagc (txa[channel].leveler.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_wcpagc (txa[channel].leveler.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].lvlrmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].lvlrmeter.p, ch[channel].dsp_size);
+    setBuffers_cfcomp (txa[channel].cfcomp.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_cfcomp (txa[channel].cfcomp.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].cfcmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].cfcmeter.p, ch[channel].dsp_size);
+    setBuffers_bandpass (txa[channel].bp0.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_bandpass (txa[channel].bp0.p, ch[channel].dsp_size);
+    setBuffers_compressor (txa[channel].compressor.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_compressor (txa[channel].compressor.p, ch[channel].dsp_size);
+    setBuffers_bandpass (txa[channel].bp1.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_bandpass (txa[channel].bp1.p, ch[channel].dsp_size);
+    setBuffers_osctrl (txa[channel].osctrl.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_osctrl (txa[channel].osctrl.p, ch[channel].dsp_size);
+    setBuffers_bandpass (txa[channel].bp2.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_bandpass (txa[channel].bp2.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].compmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].compmeter.p, ch[channel].dsp_size);
+    setBuffers_wcpagc (txa[channel].alc.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_wcpagc (txa[channel].alc.p, ch[channel].dsp_size);
+    setBuffers_ammod (txa[channel].ammod.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_ammod (txa[channel].ammod.p, ch[channel].dsp_size);
+    setBuffers_fmmod (txa[channel].fmmod.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_fmmod (txa[channel].fmmod.p, ch[channel].dsp_size);
+    setBuffers_gen (txa[channel].gen1.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_gen (txa[channel].gen1.p, ch[channel].dsp_size);
+    setBuffers_uslew (txa[channel].uslew.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_uslew (txa[channel].uslew.p, ch[channel].dsp_size);
+    setBuffers_meter (txa[channel].alcmeter.p, txa[channel].midbuff);
+    setSize_meter (txa[channel].alcmeter.p, ch[channel].dsp_size);
+    setBuffers_siphon (txa[channel].sip1.p, txa[channel].midbuff);
+    setSize_siphon (txa[channel].sip1.p, ch[channel].dsp_size);
+    setBuffers_iqc (txa[channel].iqc.p0, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_iqc (txa[channel].iqc.p0, ch[channel].dsp_size);
+    setBuffers_cfir (txa[channel].cfir.p, txa[channel].midbuff, txa[channel].midbuff);
+    setSize_cfir (txa[channel].cfir.p, ch[channel].dsp_size);
+    // output resampler
+    setBuffers_resample (txa[channel].rsmpout.p, txa[channel].midbuff, txa[channel].outbuff);
+    setSize_resample (txa[channel].rsmpout.p, ch[channel].dsp_size);
+    // output meter
+    setBuffers_meter (txa[channel].outmeter.p, txa[channel].outbuff);
+    setSize_meter (txa[channel].outmeter.p, ch[channel].dsp_outsize);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXAMode (int channel, int mode)
+{
+    if (txa[channel].mode != mode)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        txa[channel].mode = mode;
+        txa[channel].ammod.p->run   = 0;
+        txa[channel].fmmod.p->run   = 0;
+        txa[channel].preemph.p->run = 0;
+        switch (mode)
+        {
+        case TXA_AM:
+        case TXA_SAM:
+            txa[channel].ammod.p->run   = 1;
+            txa[channel].ammod.p->mode  = 0;
+            break;
+        case TXA_DSB:
+            txa[channel].ammod.p->run   = 1;
+            txa[channel].ammod.p->mode  = 1;
+            break;
+        case TXA_AM_LSB:
+        case TXA_AM_USB:
+            txa[channel].ammod.p->run   = 1;
+            txa[channel].ammod.p->mode  = 2;
+            break;
+        case TXA_FM:
+            txa[channel].fmmod.p->run   = 1;
+            txa[channel].preemph.p->run = 1;
+            break;
+        default:
+
+            break;
+        }
+        TXASetupBPFilters (channel);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetTXABandpassFreqs (int channel, double f_low, double f_high)
+{
+    if ((txa[channel].f_low != f_low) || (txa[channel].f_high != f_high))
+    {
+        txa[channel].f_low = f_low;
+        txa[channel].f_high = f_high;
+        TXASetupBPFilters (channel);
+    }
+}
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       TXA Internal Functions                                          *
+*                                                                                                       *
+********************************************************************************************************/
+
+void TXAResCheck (int channel)
+{
+    RESAMPLE a = txa[channel].rsmpin.p;
+    if (ch[channel].in_rate  != ch[channel].dsp_rate)   a->run = 1;
+    else                                                a->run = 0;
+    a = txa[channel].rsmpout.p;
+    if (ch[channel].dsp_rate != ch[channel].out_rate)   a->run = 1;
+    else                                                a->run = 0;
+}
+
+int TXAUslewCheck (int channel)
+{
+    return  (txa[channel].ammod.p->run == 1) ||
+            (txa[channel].fmmod.p->run == 1) ||
+            (txa[channel].gen0.p->run  == 1) ||
+            (txa[channel].gen1.p->run  == 1);
+}
+
+void TXASetupBPFilters (int channel)
+{
+    txa[channel].bp0.p->run = 1;
+    txa[channel].bp1.p->run = 0;
+    txa[channel].bp2.p->run = 0;
+    switch (txa[channel].mode)
+    {
+    case TXA_LSB:
+    case TXA_USB:
+    case TXA_CWL:
+    case TXA_CWU:
+    case TXA_DIGL:
+    case TXA_DIGU:
+    case TXA_SPEC:
+    case TXA_DRM:
+        CalcBandpassFilter (txa[channel].bp0.p, txa[channel].f_low, txa[channel].f_high, 2.0);
+        if (txa[channel].compressor.p->run)
+        {
+            CalcBandpassFilter (txa[channel].bp1.p, txa[channel].f_low, txa[channel].f_high, 2.0);
+            txa[channel].bp1.p->run = 1;
+            if (txa[channel].osctrl.p->run)
+            {
+                CalcBandpassFilter (txa[channel].bp2.p, txa[channel].f_low, txa[channel].f_high, 1.0);
+                txa[channel].bp2.p->run = 1;
+            }
+        }
+        break;
+    case TXA_DSB:
+    case TXA_AM:
+    case TXA_SAM:
+    case TXA_FM:
+        if (txa[channel].compressor.p->run)
+        {
+            CalcBandpassFilter (txa[channel].bp0.p, 0.0, txa[channel].f_high, 2.0);
+            CalcBandpassFilter (txa[channel].bp1.p, 0.0, txa[channel].f_high, 2.0);
+            txa[channel].bp1.p->run = 1;
+            if (txa[channel].osctrl.p->run)
+            {
+                CalcBandpassFilter (txa[channel].bp2.p, 0.0, txa[channel].f_high, 1.0);
+                txa[channel].bp2.p->run = 1;
+            }
+        }
+        else
+        {
+            CalcBandpassFilter (txa[channel].bp0.p, txa[channel].f_low, txa[channel].f_high, 1.0);
+        }
+        break;
+    case TXA_AM_LSB:
+        CalcBandpassFilter (txa[channel].bp0.p, -txa[channel].f_high, 0.0, 2.0);
+        if (txa[channel].compressor.p->run)
+        {
+            CalcBandpassFilter (txa[channel].bp1.p, -txa[channel].f_high, 0.0, 2.0);
+            txa[channel].bp1.p->run = 1;
+            if (txa[channel].osctrl.p->run)
+            {
+                CalcBandpassFilter (txa[channel].bp2.p, -txa[channel].f_high, 0.0, 1.0);
+                txa[channel].bp2.p->run = 1;
+            }
+        }
+        break;
+    case TXA_AM_USB:
+        CalcBandpassFilter (txa[channel].bp0.p, 0.0, txa[channel].f_high, 2.0);
+        if (txa[channel].compressor.p->run)
+        {
+            CalcBandpassFilter (txa[channel].bp1.p, 0.0, txa[channel].f_high, 2.0);
+            txa[channel].bp1.p->run = 1;
+            if (txa[channel].osctrl.p->run)
+            {
+                CalcBandpassFilter (txa[channel].bp2.p, 0.0, txa[channel].f_high, 1.0);
+                txa[channel].bp2.p->run = 1;
+            }
+        }
+        break;
+    }
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                               Collectives                                             *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void TXASetNC (int channel, int nc)
+{
+    int oldstate = SetChannelState (channel, 0, 1);
+    SetTXABandpassNC            (channel, nc);
+    SetTXAFMEmphNC              (channel, nc);
+    SetTXAEQNC                  (channel, nc);
+    SetTXAFMNC                  (channel, nc);
+    SetTXACFIRNC                (channel, nc);
+    SetChannelState (channel, oldstate, 0);
+}
+
+PORT
+void TXASetMP (int channel, int mp)
+{
+    SetTXABandpassMP            (channel, mp);
+    SetTXAFMEmphMP              (channel, mp);
+    SetTXAEQMP                  (channel, mp);
+    SetTXAFMMP                  (channel, mp);
+}
+
+PORT
+void SetTXAFMAFFilter (int channel, double low, double high)
+{
+    SetTXAFMPreEmphFreqs (channel, low, high);
+    SetTXAFMAFFreqs (channel, low, high);
+}

wdsp/TXA.h

@@ -0,0 +1,193 @@
+/*  TXA.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2017 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#ifndef _txa_h
+#define _txa_h
+#include "comm.h"
+
+enum txaMode
+{
+    TXA_LSB,
+    TXA_USB,
+    TXA_DSB,
+    TXA_CWL,
+    TXA_CWU,
+    TXA_FM,
+    TXA_AM,
+    TXA_DIGU,
+    TXA_SPEC,
+    TXA_DIGL,
+    TXA_SAM,
+    TXA_DRM,
+    TXA_AM_LSB,
+    TXA_AM_USB
+};
+
+enum txaMeterType
+{
+    TXA_MIC_PK,
+    TXA_MIC_AV,
+    TXA_EQ_PK,
+    TXA_EQ_AV,
+    TXA_LVLR_PK,
+    TXA_LVLR_AV,
+    TXA_LVLR_GAIN,
+    TXA_CFC_PK,
+    TXA_CFC_AV,
+    TXA_CFC_GAIN,
+    TXA_COMP_PK,
+    TXA_COMP_AV,
+    TXA_ALC_PK,
+    TXA_ALC_AV,
+    TXA_ALC_GAIN,
+    TXA_OUT_PK,
+    TXA_OUT_AV,
+    TXA_METERTYPE_LAST
+};
+
+struct _txa
+{
+    double* inbuff;
+    double* outbuff;
+    double* midbuff;
+    int mode;
+    double f_low;
+    double f_high;
+    double meter[TXA_METERTYPE_LAST];
+    CRITICAL_SECTION* pmtupdate[TXA_METERTYPE_LAST];
+    struct
+    {
+        METER p;
+    } micmeter, eqmeter, lvlrmeter, cfcmeter, compmeter, alcmeter, outmeter;
+    struct
+    {
+        RESAMPLE p;
+    } rsmpin, rsmpout;
+    struct
+    {
+        PANEL p;
+    } panel;
+    struct
+    {
+        AMSQ p;
+    } amsq;
+    struct
+    {
+        EQP p;
+    } eqp;
+    struct
+    {
+        PHROT p;
+    } phrot;
+    struct
+    {
+        CFCOMP p;
+    } cfcomp;
+    struct
+    {
+        COMPRESSOR p;
+    } compressor;
+    struct
+    {
+        BANDPASS p;
+    } bp0, bp1, bp2;
+    struct
+    {
+        OSCTRL p;
+    } osctrl;
+    struct
+    {
+        WCPAGC p;
+    } leveler, alc;
+    struct
+    {
+        AMMOD p;
+    } ammod;
+    struct
+    {
+        EMPHP p;
+    } preemph;
+    struct
+    {
+        FMMOD p;
+    } fmmod;
+    struct
+    {
+        SIPHON p;
+    } sip1;
+    struct
+    {
+        GEN p;
+    } gen0, gen1;
+    struct
+    {
+        USLEW p;
+    } uslew;
+    struct
+    {
+        CALCC p;
+        CRITICAL_SECTION cs_update;
+    } calcc;
+    struct
+    {
+        IQC p0, p1;
+        // p0 for dsp-synchronized reference, p1 for other
+    } iqc;
+    struct
+    {
+        CFIR p;
+    } cfir;
+};
+
+extern struct _txa txa[];
+
+extern void create_txa (int channel);
+
+extern void destroy_txa (int channel);
+
+extern void flush_txa (int channel);
+
+extern void xtxa (int channel);
+
+extern int TXAUslewCheck (int channel);
+
+extern void setInputSamplerate_txa (int channel);
+
+extern void setOutputSamplerate_txa (int channel);
+
+extern void setDSPSamplerate_txa (int channel);
+
+extern void setDSPBuffsize_txa (int channel);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXAMode (int channel, int mode);
+
+extern void TXAResCheck (int channel);
+
+extern void TXASetupBPFilters (int channel);
+
+#endif

wdsp/amd.c

@@ -0,0 +1,293 @@
+/*  amd.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2012, 2013 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#include "comm.h"
+
+AMD create_amd
+    (
+    int run,
+    int buff_size,
+    double *in_buff,
+    double *out_buff,
+    int mode,
+    int levelfade,
+    int sbmode,
+    int sample_rate,
+    double fmin,
+    double fmax,
+    double zeta,
+    double omegaN,
+    double tauR,
+    double tauI
+    )
+{
+    AMD a = (AMD) malloc0 (sizeof(amd));
+    a->run = run;
+    a->buff_size = buff_size;
+    a->in_buff = in_buff;
+    a->out_buff = out_buff;
+    a->mode = mode;
+    a->levelfade = levelfade;
+    a->sbmode = sbmode;
+    a->sample_rate = (double)sample_rate;
+    a->fmin = fmin;
+    a->fmax = fmax;
+    a->zeta = zeta;
+    a->omegaN = omegaN;
+    a->tauR = tauR;
+    a->tauI = tauI;
+
+    init_amd(a);
+    return a;
+}
+
+void destroy_amd(AMD a)
+{
+    _aligned_free (a);
+}
+
+void init_amd(AMD a)
+{
+    //pll
+    a->omega_min = TWOPI * a->fmin / a->sample_rate;
+    a->omega_max = TWOPI * a->fmax / a->sample_rate;
+    a->g1 = 1.0 - exp(-2.0 * a->omegaN * a->zeta / a->sample_rate);
+    a->g2 = -a->g1 + 2.0 * (1 - exp(-a->omegaN * a->zeta / a->sample_rate) * cos(a->omegaN / a->sample_rate * sqrt(1.0 - a->zeta * a->zeta)));
+    a->phs = 0.0;
+    a->fil_out = 0.0;
+    a->omega = 0.0;
+
+    //fade leveler
+    a->dc = 0.0;
+    a->dc_insert = 0.0;
+    a->mtauR = exp(-1.0 / (a->sample_rate * a->tauR));
+    a->onem_mtauR = 1.0 - a->mtauR;
+    a->mtauI = exp(-1.0 / (a->sample_rate * a->tauI));
+    a->onem_mtauI = 1.0 - a->mtauI;
+
+    //sideband separation
+    a->c0[0] = -0.328201924180698;
+    a->c0[1] = -0.744171491539427;
+    a->c0[2] = -0.923022915444215;
+    a->c0[3] = -0.978490468768238;
+    a->c0[4] = -0.994128272402075;
+    a->c0[5] = -0.998458978159551;
+    a->c0[6] = -0.999790306259206;
+
+    a->c1[0] = -0.0991227952747244;
+    a->c1[1] = -0.565619728761389;
+    a->c1[2] = -0.857467122550052;
+    a->c1[3] = -0.959123933111275;
+    a->c1[4] = -0.988739372718090;
+    a->c1[5] = -0.996959189310611;
+    a->c1[6] = -0.999282492800792;
+}
+
+void flush_amd (AMD a)
+{
+    a->dc = 0.0;
+    a->dc_insert = 0.0;
+}
+
+void xamd (AMD a)
+{
+    int i;
+    double audio;
+    double vco[2];
+    double corr[2];
+    double det;
+    double del_out;
+    double ai, bi, aq, bq;
+    double ai_ps, bi_ps, aq_ps, bq_ps;
+    int j, k;
+    if (a->run)
+    {
+        switch (a->mode)
+        {
+
+            case 0:     //AM Demodulator
+                {
+                    for (i = 0; i < a->buff_size; i++)
+                    {
+                        audio = sqrt(a->in_buff[2 * i + 0] * a->in_buff[2 * i + 0] + a->in_buff[2 * i + 1] * a->in_buff[2 * i + 1]);
+                        if (a->levelfade)
+                        {
+                            a->dc = a->mtauR * a->dc + a->onem_mtauR * audio;
+                            a->dc_insert = a->mtauI * a->dc_insert + a->onem_mtauI * audio;
+                            audio += a->dc_insert - a->dc;
+                        }
+                        a->out_buff[2 * i + 0] = audio;
+                        a->out_buff[2 * i + 1] = audio;
+                    }
+                    break;
+                }
+
+            case 1:     //Synchronous AM Demodulator with Sideband Separation
+                {
+                    for (i = 0; i < a->buff_size; i++)
+                    {
+                        vco[0] = cos(a->phs);
+                        vco[1] = sin(a->phs);
+
+                        ai = a->in_buff[2 * i + 0] * vco[0];
+                        bi = a->in_buff[2 * i + 0] * vco[1];
+                        aq = a->in_buff[2 * i + 1] * vco[0];
+                        bq = a->in_buff[2 * i + 1] * vco[1];
+
+                        if (a->sbmode != 0)
+                        {
+                            a->a[0] = a->dsI;
+                            a->b[0] = bi;
+                            a->c[0] = a->dsQ;
+                            a->d[0] = aq;
+                            a->dsI = ai;
+                            a->dsQ = bq;
+
+                            for (j = 0; j < STAGES; j++)
+                            {
+                                k = 3 * j;
+                                a->a[k + 3] = a->c0[j] * (a->a[k] - a->a[k + 5]) + a->a[k + 2];
+                                a->b[k + 3] = a->c1[j] * (a->b[k] - a->b[k + 5]) + a->b[k + 2];
+                                a->c[k + 3] = a->c0[j] * (a->c[k] - a->c[k + 5]) + a->c[k + 2];
+                                a->d[k + 3] = a->c1[j] * (a->d[k] - a->d[k + 5]) + a->d[k + 2];
+                            }
+                            ai_ps = a->a[OUT_IDX];
+                            bi_ps = a->b[OUT_IDX];
+                            bq_ps = a->c[OUT_IDX];
+                            aq_ps = a->d[OUT_IDX];
+
+                            for (j = OUT_IDX + 2; j > 0; j--)
+                            {
+                                a->a[j] = a->a[j - 1];
+                                a->b[j] = a->b[j - 1];
+                                a->c[j] = a->c[j - 1];
+                                a->d[j] = a->d[j - 1];
+                            }
+                        }
+
+                        corr[0] = +ai + bq;
+                        corr[1] = -bi + aq;
+
+                        switch(a->sbmode)
+                        {
+                        case 0: //both sidebands
+                            {
+                                audio = corr[0];
+                                break;
+                            }
+                        case 1: //LSB
+                            {
+                                audio = (ai_ps - bi_ps) + (aq_ps + bq_ps);
+                                break;
+                            }
+                        case 2: //USB
+                            {
+                                audio = (ai_ps + bi_ps) - (aq_ps - bq_ps);
+                                break;
+                            }
+                        }
+
+                        if (a->levelfade)
+                        {
+                            a->dc = a->mtauR * a->dc + a->onem_mtauR * audio;
+                            a->dc_insert = a->mtauI * a->dc_insert + a->onem_mtauI * corr[0];
+                            audio += a->dc_insert - a->dc;
+                        }
+                        a->out_buff[2 * i + 0] = audio;
+                        a->out_buff[2 * i + 1] = audio;
+
+                        if ((corr[0] == 0.0) && (corr[1] == 0.0)) corr[0] = 1.0;
+                        det = atan2(corr[1], corr[0]);
+                        del_out = a->fil_out;
+                        a->omega += a->g2 * det;
+                        if (a->omega < a->omega_min) a->omega = a->omega_min;
+                        if (a->omega > a->omega_max) a->omega = a->omega_max;
+                        a->fil_out = a->g1 * det + a->omega;
+                        a->phs += del_out;
+                        while (a->phs >= TWOPI) a->phs -= TWOPI;
+                        while (a->phs < 0.0) a->phs += TWOPI;
+                    }
+                    break;
+                }
+        }
+    }
+    else if (a->in_buff != a->out_buff)
+        memcpy (a->out_buff, a->in_buff, a->buff_size * sizeof(complex));
+}
+
+void setBuffers_amd (AMD a, double* in, double* out)
+{
+    a->in_buff = in;
+    a->out_buff = out;
+}
+
+void setSamplerate_amd (AMD a, int rate)
+{
+    a->sample_rate = rate;
+    init_amd(a);
+}
+
+void setSize_amd (AMD a, int size)
+{
+    a->buff_size = size;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetRXAAMDRun(int channel, int run)
+{
+    AMD a = rxa[channel].amd.p;
+    if (a->run != run)
+    {
+        RXAbp1Check (channel, run, rxa[channel].snba.p->run, rxa[channel].emnr.p->run,
+            rxa[channel].anf.p->run, rxa[channel].anr.p->run);
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->run = run;
+        RXAbp1Set (channel);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT void
+SetRXAAMDSBMode(int channel, int sbmode)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].amd.p->sbmode = sbmode;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAAMDFadeLevel(int channel, int levelfade)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].amd.p->levelfade = levelfade;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/amd.h

@@ -0,0 +1,118 @@
+/*  amd.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2012, 2013 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#ifndef _amd_h
+#define _amd_h
+
+// ff defines for sbdemod
+#ifndef STAGES
+#define STAGES      7
+#endif
+
+#ifndef OUT_IDX
+#define OUT_IDX     (3 * STAGES)
+#endif
+
+typedef struct _amd
+{
+    int run;
+    int buff_size;                      // buffer size
+    double *in_buff;                    // pointer to input buffer
+    double *out_buff;                   // pointer to output buffer
+    int mode;                           // demodulation mode
+    double sample_rate;                 // sample rate
+    double dc;                          // dc component in demodulated output
+    double fmin;                        // pll - minimum carrier freq to lock
+    double fmax;                        // pll - maximum carrier freq to lock
+    double omega_min;                   // pll - minimum lock check parameter
+    double omega_max;                   // pll - maximum lock check parameter
+    double zeta;                        // pll - damping factor; as coded, must be <=1.0
+    double omegaN;                      // pll - natural frequency
+    double phs;                         // pll - phase accumulator
+    double omega;                       // pll - locked pll frequency
+    double fil_out;                     // pll - filter output
+    double g1, g2;                      // pll - filter gain parameters
+    double tauR;                        // carrier removal time constant
+    double tauI;                        // carrier insertion time constant
+    double mtauR;                       // carrier removal multiplier
+    double onem_mtauR;                  // 1.0 - carrier_removal_multiplier
+    double mtauI;                       // carrier insertion multiplier
+    double onem_mtauI;                  // 1.0 - carrier_insertion_multiplier
+    double a[3 * STAGES + 3];           // Filter a variables
+    double b[3 * STAGES + 3];           // Filter b variables
+    double c[3 * STAGES + 3];           // Filter c variables
+    double d[3 * STAGES + 3];           // Filter d variables
+    double c0[STAGES];                  // Filter coefficients - path 0
+    double c1[STAGES];                  // Filter coefficients - path 1
+    double dsI;                         // delayed sample, I path
+    double dsQ;                         // delayed sample, Q path
+    double dc_insert;                   // dc component to insert in output
+    int sbmode;                         // sideband mode
+    int levelfade;                      // Fade Leveler switch
+
+}amd, *AMD;
+
+extern AMD create_amd
+    (
+    int run,
+    int buff_size,
+    double *in_buff,
+    double *out_buff,
+    int mode,
+    int levelfade,
+    int sbmode,
+    int sample_rate,
+    double fmin,
+    double fmax,
+    double zeta,
+    double omegaN,
+    double tauR,
+    double tauI
+    );
+
+extern void init_amd (AMD a);
+
+extern void destroy_amd (AMD a);
+
+extern void flush_amd (AMD a);
+
+extern void xamd (AMD a);
+
+extern void setBuffers_amd (AMD a, double* in, double* out);
+
+extern void setSamplerate_amd (AMD a, int rate);
+
+extern void setSize_amd (AMD a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAAMDRun(int channel, int run);
+
+extern __declspec (dllexport) void SetRXAAMDSBMode(int channel, int sbmode);
+
+extern __declspec (dllexport) void SetRXAAMDFadeLevel(int channel, int levelfade);
+
+#endif

wdsp/ammod.c

@@ -0,0 +1,110 @@
+/*  ammod.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2017 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#include "comm.h"
+
+AMMOD create_ammod (int run, int mode, int size, double* in, double* out, double c_level)
+{
+    AMMOD a = (AMMOD) malloc0 (sizeof (ammod));
+    a->run = run;
+    a->mode = mode;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->c_level = c_level;
+    a->a_level = 1.0 - a->c_level;
+    a->mult = 1.0 / sqrt (2.0);
+    return a;
+}
+
+void destroy_ammod (AMMOD a)
+{
+    _aligned_free (a);
+}
+
+void flush_ammod (AMMOD a)
+{
+
+}
+
+void xammod (AMMOD a)
+{
+    if (a->run)
+    {
+        int i;
+        switch (a->mode)
+        {
+        case 0: // AM
+            for (i = 0; i < a->size; i++)
+                a->out[2 * i + 0] = a->out[2 * i + 1] = a->mult * (a->c_level + a->a_level * a->in[2 * i + 0]);
+            break;
+        case 1: // DSB
+            for (i = 0; i < a->size; i++)
+                a->out[2 * i + 0] = a->out[2 * i + 1] = a->mult * a->in[2 * i + 0];
+            break;
+        case 2: // SSB w/Carrier
+            for (i = 0; i < a->size; i++)
+            {
+                a->out[2 * i + 0] = a->mult * a->c_level + a->a_level * a->in[2 * i + 0];
+                a->out[2 * i + 1] = a->mult * a->c_level + a->a_level * a->in[2 * i + 1];
+            }
+            break;
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_ammod (AMMOD a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+}
+
+void setSamplerate_ammod (AMMOD a, int rate)
+{
+
+}
+
+void setSize_ammod (AMMOD a, int size)
+{
+    a->size = size;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetTXAAMCarrierLevel (int channel, double c_level)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].ammod.p->c_level = c_level;
+    txa[channel].ammod.p->a_level = 1.0 - c_level;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/ammod.h

@@ -0,0 +1,60 @@
+/*  ammod.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#ifndef _ammod_h
+#define _ammod_h
+
+typedef struct _ammod
+{
+    int run;
+    int mode;
+    int size;
+    double* in;
+    double* out;
+    double c_level;
+    double a_level;
+    double mult;
+}ammod, *AMMOD;
+
+extern AMMOD create_ammod (int run, int mode, int size, double* in, double* out, double c_level);
+
+extern void destroy_ammod (AMMOD a);
+
+extern void flush_ammod (AMMOD a);
+
+extern void xammod (AMMOD a);
+
+extern void setBuffers_ammod (AMMOD a, double* in, double* out);
+
+extern void setSamplerate_ammod (AMMOD a, int rate);
+
+extern void setSize_ammod (AMMOD a, int size);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXAAMCarrierLevel (int channel, double c_level);
+
+#endif

wdsp/amsq.c

@@ -0,0 +1,272 @@
+/*  amsq.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+
+#include "comm.h"
+
+void compute_slews(AMSQ a)
+{
+    int i;
+    double delta, theta;
+    delta = PI / (double)a->ntup;
+    theta = 0.0;
+    for (i = 0; i <= a->ntup; i++)
+    {
+        a->cup[i] = a->muted_gain + (1.0 - a->muted_gain) * 0.5 * (1.0 - cos (theta));
+        theta += delta;
+    }
+    delta = PI / (double)a->ntdown;
+    theta = 0.0;
+    for (i = 0; i <= a->ntdown; i++)
+    {
+        a->cdown[i] = a->muted_gain + (1.0 - a->muted_gain) * 0.5 * (1.0 + cos (theta));
+        theta += delta;
+    }
+}
+
+void calc_amsq(AMSQ a)
+{
+    // signal averaging
+    a->trigsig = (double *)malloc0(a->size * sizeof(complex));
+    a->avm = exp(-1.0 / (a->rate * a->avtau));
+    a->onem_avm = 1.0 - a->avm;
+    a->avsig = 0.0;
+    // level change
+    a->ntup = (int)(a->tup * a->rate);
+    a->ntdown = (int)(a->tdown * a->rate);
+    a->cup = (double *)malloc0((a->ntup + 1) * sizeof(double));
+    a->cdown = (double *)malloc0((a->ntdown + 1) * sizeof(double));
+    compute_slews(a);
+    // control
+    a->state = 0;
+}
+
+void decalc_amsq (AMSQ a)
+{
+    _aligned_free (a->cdown);
+    _aligned_free (a->cup);
+    _aligned_free (a->trigsig);
+}
+
+AMSQ create_amsq (int run, int size, double* in, double* out, double* trigger, int rate, double avtau,
+    double tup, double tdown, double tail_thresh, double unmute_thresh, double min_tail, double max_tail, double muted_gain)
+{
+    AMSQ a = (AMSQ) malloc0 (sizeof (amsq));
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = (double)rate;
+    a->muted_gain = muted_gain;
+    a->trigger = trigger;
+    a->avtau = avtau;
+    a->tup = tup;
+    a->tdown = tdown;
+    a->tail_thresh = tail_thresh;
+    a->unmute_thresh = unmute_thresh;
+    a->min_tail = min_tail;
+    a->max_tail = max_tail;
+    calc_amsq (a);
+    return a;
+}
+
+void destroy_amsq (AMSQ a)
+{
+    decalc_amsq (a);
+    _aligned_free (a);
+}
+
+void flush_amsq (AMSQ a)
+{
+    memset (a->trigsig, 0, a->size * sizeof (complex));
+    a->avsig = 0.0;
+    a->state = 0;
+}
+
+enum _amsqstate
+{
+    MUTED,
+    INCREASE,
+    UNMUTED,
+    TAIL,
+    DECREASE
+};
+
+void xamsq (AMSQ a)
+{
+    if (a->run)
+    {
+        int i;
+        double sig, siglimit;
+        for (i = 0; i < a->size; i++)
+        {
+            sig = sqrt (a->trigsig[2 * i + 0] * a->trigsig[2 * i + 0] + a->trigsig[2 * i + 1] * a->trigsig[2 * i + 1]);
+            a->avsig = a->avm * a->avsig + a->onem_avm * sig;
+            switch (a->state)
+            {
+            case MUTED:
+                if (a->avsig > a->unmute_thresh)
+                {
+                    a->state = INCREASE;
+                    a->count = a->ntup;
+                }
+                a->out[2 * i + 0] = a->muted_gain * a->in[2 * i + 0];
+                a->out[2 * i + 1] = a->muted_gain * a->in[2 * i + 1];
+                break;
+            case INCREASE:
+                a->out[2 * i + 0] = a->in[2 * i + 0] * a->cup[a->ntup - a->count];
+                a->out[2 * i + 1] = a->in[2 * i + 1] * a->cup[a->ntup - a->count];
+                if (a->count-- == 0)
+                    a->state = UNMUTED;
+                break;
+            case UNMUTED:
+                if (a->avsig < a->tail_thresh)
+                {
+                    a->state = TAIL;
+                    if ((siglimit = a->avsig) > 1.0) siglimit = 1.0;
+                    a->count = (int)((a->min_tail + (a->max_tail - a->min_tail) * (1.0 - siglimit)) * a->rate);
+                }
+                a->out[2 * i + 0] = a->in[2 * i + 0];
+                a->out[2 * i + 1] = a->in[2 * i + 1];
+                break;
+            case TAIL:
+                a->out[2 * i + 0] = a->in[2 * i + 0];
+                a->out[2 * i + 1] = a->in[2 * i + 1];
+                if (a->avsig > a->unmute_thresh)
+                    a->state = UNMUTED;
+                else if (a->count-- == 0)
+                {
+                    a->state = DECREASE;
+                    a->count = a->ntdown;
+                }
+                break;
+            case DECREASE:
+                a->out[2 * i + 0] = a->in[2 * i + 0] * a->cdown[a->ntdown - a->count];
+                a->out[2 * i + 1] = a->in[2 * i + 1] * a->cdown[a->ntdown - a->count];
+                if (a->count-- == 0)
+                    a->state = MUTED;
+                break;
+            }
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void xamsqcap (AMSQ a)
+{
+    memcpy (a->trigsig, a->trigger, a->size * sizeof (complex));
+}
+
+void setBuffers_amsq (AMSQ a, double* in, double* out, double* trigger)
+{
+    a->in = in;
+    a->out = out;
+    a->trigger = trigger;
+}
+
+void setSamplerate_amsq (AMSQ a, int rate)
+{
+    decalc_amsq (a);
+    a->rate = rate;
+    calc_amsq (a);
+}
+
+void setSize_amsq (AMSQ a, int size)
+{
+    decalc_amsq (a);
+    a->size = size;
+    calc_amsq (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXAAMSQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].amsq.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAAMSQThreshold (int channel, double threshold)
+{
+    double thresh = pow (10.0, threshold / 20.0);
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].amsq.p->tail_thresh = 0.9 * thresh;
+    rxa[channel].amsq.p->unmute_thresh =  thresh;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAAMSQMaxTail (int channel, double tail)
+{
+    AMSQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].amsq.p;
+    if (tail < a->min_tail) tail = a->min_tail;
+    a->max_tail = tail;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXAAMSQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].amsq.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAAMSQMutedGain (int channel, double dBlevel)
+{   // dBlevel is negative
+    AMSQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].amsq.p;
+    a->muted_gain = pow (10.0, dBlevel / 20.0);
+    compute_slews(a);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAAMSQThreshold (int channel, double threshold)
+{
+    double thresh = pow (10.0, threshold / 20.0);
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].amsq.p->tail_thresh = 0.9 * thresh;
+    txa[channel].amsq.p->unmute_thresh =  thresh;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/amsq.h

@@ -0,0 +1,89 @@
+/*  amsq.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013 Warren Pratt, NR0V
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+
+The author can be reached by email at
+
+warren@wpratt.com
+
+*/
+#ifndef _amsq_h
+#define _amsq_h
+
+typedef struct _amsq
+{
+    int run;                            // 0 if squelch system is OFF; 1 if it's ON
+    int size;                           // size of input/output buffers
+    double* in;                         // squelch input signal buffer
+    double* out;                        // squelch output signal buffer
+    double* trigger;                    // pointer to trigger data source
+    double* trigsig;                    // buffer containing trigger signal
+    double rate;                        // sample rate
+    double avtau;                       // time constant for averaging noise
+    double avm;
+    double onem_avm;
+    double avsig;
+    int state;                          // state machine control
+    int count;
+    double tup;
+    double tdown;
+    int ntup;
+    int ntdown;
+    double* cup;
+    double* cdown;
+    double tail_thresh;
+    double unmute_thresh;
+    double min_tail;
+    double max_tail;
+    double muted_gain;
+} amsq, *AMSQ;
+
+extern AMSQ create_amsq (int run, int size, double* in, double* out, double* trigger, int rate, double avtau, double tup, double tdown, double tail_thresh, double unmute_thresh, double min_tail, double max_tail, double muted_gain);
+
+extern void destroy_amsq (AMSQ a);
+
+extern void flush_amsq (AMSQ a);
+
+extern void xamsq (AMSQ a);
+
+extern void xamsqcap (AMSQ a);
+
+extern void setBuffers_amsq (AMSQ a, double* in, double* out, double* trigger);
+
+extern void setSamplerate_amsq (AMSQ a, int rate);
+
+extern void setSize_amsq (AMSQ a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAAMSQRun (int channel, int run);
+
+extern __declspec (dllexport) void SetRXAAMSQThreshold (int channel, double threshold);
+
+extern __declspec (dllexport) void SetRXAAMSQMaxTail (int channel, double tail);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXAAMSQRun (int channel, int run);
+
+extern __declspec (dllexport) void SetTXAAMSQMutedGain (int channel, double dBlevel);
+
+extern __declspec (dllexport) void SetTXAAMSQThreshold (int channel, double threshold);
+
+#endif