Quisk version 4.2.49

CHANGELOG.txt

@@ -1,3 +1,182 @@
+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

@@ -1,5 +1,5 @@
 # Quisk Software Defined Radio by N2ADR
-**December 26, 2023**
+**December 27, 2023**
 
 This is Quisk, a Software Defined Radio (SDR).
 You supply radio hardware such as Hermes Lite2 or SoftRock to convert
@@ -14,8 +14,8 @@ The web page for this project is  [https://groups.io/g/n2adr-sdr](https://groups
 
 The change log is [here](CHANGELOG.txt).
 
-The documentation is [here](docs.html).
+Installation instructions are [here](http://james.ahlstrom.name/quisk/docs.html#Installation).
 
-The help file is [here](help.html).
+The documentation is [here](http://james.ahlstrom.name/quisk/docs.html).
 
-This site is under construction. Check back soon.
+The help file is [here](http://james.ahlstrom.name/quisk/help.html).

__init__.py

@@ -1,2 +1,2 @@
-#Quisk version 4.2.29
+#Quisk version 4.2.49
 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)

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 (September 2025)
 </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>
@@ -189,23 +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.
+<b>On Windows Quisk requires Python 3.8, 3.9, 3.10, 3.11, 3.12 or 3.13.</b>
+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>
+<b>On Windows Quisk requires Python 3.8, 3.9, 3.10, 3.11, 3.12 or 3.13.</b>
 There may be newer versions of Python, but until support is added, use the versions listed above.
 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
@@ -240,17 +243,24 @@ 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".
 </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 +290,112 @@ 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:
+</p>
+<br>
+<pre>
+cd ~/quisk
+git pull
+</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.
+<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 +538,7 @@ 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 +836,14 @@ 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 +852,20 @@ 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>
@@ -1544,28 +1491,47 @@ 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 +1607,46 @@ 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

filters.h

@@ -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,

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.

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):

is_key_down.c

@@ -168,6 +168,7 @@ 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

microphone.c

@@ -8,7 +8,6 @@
 #include <sys/types.h>
 #include "microphone.h"
 #include "filter.h"
-#include "freedv.h"
 
 #ifdef MS_WINDOWS
 #include <winsock2.h>
@@ -75,6 +74,8 @@ 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 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 +244,9 @@ 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 +371,7 @@ 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 +422,10 @@ 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 +541,8 @@ 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 +573,8 @@ 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 +614,7 @@ 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 +623,6 @@ 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 +759,8 @@ 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 +776,7 @@ 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 +784,7 @@ 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 +797,7 @@ 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 +841,33 @@ 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 +916,33 @@ 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 +1008,47 @@ 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;
@@ -1301,6 +1260,8 @@ 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 +1269,6 @@ 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;
 		}
@@ -1403,11 +1359,6 @@ 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 +1380,31 @@ 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 +1528,6 @@ 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 +1573,527 @@ 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/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'

quisk.c

@@ -5,6 +5,7 @@
 #include <complex.h>	// Use native C99 complex type for fftw3
 #include <fftw3.h>
 #include <sys/types.h>
+#include <stdbool.h>
 
 #ifdef MS_WINDOWS
 #include <winsock2.h>
@@ -107,7 +108,7 @@ 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
@@ -150,9 +151,14 @@ 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 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 +185,7 @@ 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 +218,17 @@ 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 +571,10 @@ 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 +1850,6 @@ 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 +1922,10 @@ 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 +1945,10 @@ 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 +1971,7 @@ 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 +2085,7 @@ 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 +2112,7 @@ 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 +2151,6 @@ 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 +2373,7 @@ 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 +2445,8 @@ 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 +2648,7 @@ 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 +2658,7 @@ 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 +2686,6 @@ 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 +2843,8 @@ 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 +2853,6 @@ 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);
@@ -3084,6 +3058,10 @@ 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;
 }
@@ -3113,11 +3091,12 @@ 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", 
+	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, "|iiiiiiiidii", kwlist,
 	&quisk_is_vna, &nbytes, &py_sample_rx_endian, &read_error, &clip,
 	&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause))
 		return NULL;
@@ -3160,31 +3139,43 @@ 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 +3676,7 @@ 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 +3723,24 @@ 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 +3763,43 @@ 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 +3830,10 @@ 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 +3939,16 @@ 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
@@ -4168,6 +4216,13 @@ 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
 
@@ -4629,6 +4684,9 @@ 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 +5010,135 @@ 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 +5211,10 @@ 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 +5265,9 @@ 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
@@ -5786,6 +5979,7 @@ 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 +6093,7 @@ 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."},
+	{"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 +6140,7 @@ 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 +6181,6 @@ 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."},

quisk.h

@@ -312,13 +312,11 @@ 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 +335,14 @@ 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 +352,7 @@ 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 +386,6 @@ 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 +393,7 @@ 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);
 
 // Driver function definitions=================================================
 int  quisk_read_alsa(struct sound_dev *, complex double *);
@@ -464,7 +450,7 @@ 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 +465,7 @@ 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 +1407,31 @@ 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'
-
-## 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 = ""
-
 ## 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 +1482,64 @@ 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 +1804,15 @@ 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 +1911,6 @@ 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 +1920,7 @@ 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 +2040,8 @@ 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 +2049,7 @@ color_scheme = 'A'
 waterfall_palette = 'A'
 #waterfall_palette = 'B'
 #waterfall_palette = 'C'
+#waterfall_palette = 'DK1MI_AM'
 
 
 
@@ -1931,6 +2084,8 @@ 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 +2115,8 @@ 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 +2154,73 @@ 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 +2278,18 @@ 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 +2315,26 @@ 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

@@ -120,6 +120,8 @@ class Hardware:
     self.application.bandscope_clock = clock
   #def PollCwKey(self):  # Optional. Called frequently by the sound thread to check the CW key status.
   #  pass        # Do not define if not needed.
+  #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 sending samples.
     # If you return a string, it replaces the string returned from hardware open()
     pass

quisk_wdsp.c

@@ -2,60 +2,68 @@
 
 #include <Python.h>
 #include <stdlib.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)

setup.py

@@ -7,7 +7,7 @@ import os
 # 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'
+Version = '4.2.49'
 
 fp = open("__init__.py", "w")	# write title string
 fp.write("#Quisk version %s\n" % Version)
@@ -39,8 +39,10 @@ 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"):
@@ -83,7 +85,7 @@ 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']},
 	entry_points = {'gui_scripts' : ['quisk = quisk.quisk:main', 'quisk_vna = quisk.quisk_vna:main']},

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

@@ -85,8 +85,10 @@ 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;
@@ -175,9 +177,10 @@ 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 +733,11 @@ 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 +842,34 @@ 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 +947,10 @@ 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
@@ -965,21 +1000,25 @@ 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)
+	//if (is_DGT)
 		play_sound_interface(&DigitalOutput, nSamples, cSamples, 1, digital_output_level);
    
 	// Perhaps record the speaker audio to a file
@@ -1075,7 +1114,7 @@ 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 +1440,8 @@ 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 +1606,9 @@ 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

@@ -495,8 +495,37 @@ 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 +544,6 @@ 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);

sound_pulseaudio.c

@@ -64,12 +64,20 @@ 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. 
  */
@@ -1043,19 +1051,23 @@ 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) {
@@ -1070,6 +1082,9 @@ 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 +1128,41 @@ 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;

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
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+
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+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,
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+
+Activities other than copying, distribution and modification are not
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+
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+infringement or for any other reason (not limited to patent issues),
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+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
+apply and the section as a whole is intended to apply in other
+circumstances.
+
+It is not the purpose of this section to induce you to infringe any
+patents or other property right claims or to contest validity of any
+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
+through that system in reliance on consistent application of that
+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
+MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.  THE ENTIRE RISK AS
+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,
+INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
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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
+compress.o: nobII.h osctrl.h patchpanel.h resample.h rmatch.h varsamp.h RXA.h
+compress.o: sender.h shift.h siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h
+compress.o: utilities.h
+delay.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
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+delay.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
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+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

wdsp/analyzer.h

@@ -0,0 +1,195 @@
+/*  analyzer.h
+
+This file is part of a program that implements a Spectrum Analyzer
+used in conjunction with software-defined-radio hardware.
+
+Copyright (C) 2012, 2013, 2014, 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
+
+*/
+#ifndef _analyzer_h
+#define _analyzer_h
+#include "comm.h"
+
+typedef struct _dp
+{
+    int max_size;                                           // maximum fft size to be used
+    int max_num_fft;                                        // maximum number of LO positions per sub-span to be used
+    int max_stitch;                                         // maximum number of sub-spans to be concatenated
+                                                            //      NOTE:  max_size, max_num_fft, and max_stitch MUST BE <= THE
+                                                            //      CORRESPONDING VALUES IN <analyzer.h>!!
+    int num_fft;                                            // current number of ffts in use
+    int num_pixout;                                         // current number of detector/averages/pixel value outputs
+    int size;                                               // current size of fft input sample vector
+    int out_size;                                           // current size of fft output vector
+    int window_type;                                        // type of the window function to be applied
+    int overlap;                                            // number of samples re-used per fft, range 0 to size-1
+    int flip[dMAX_NUM_FFT];                                 // 0 for low-side LO => do NOT flip; 1 for high-side LO => FLIP
+    int clip;                                               // number of bins to clip off on EACH end of the sub-span fft
+                                                            //      ASSUMES size/2 IS AN EVEN NUMBER!!!
+    double fsclipL;                                         // number of intervals to clip off the lower end of the TOTAL SPAN
+    double fsclipH;                                         // number of intervals to clip off the upper end of the TOTAL SPAN
+    int fscL;                                               // fsclipL modulo (out_size - 2 * clip)
+    int fscH;                                               // fsclipH modulo (out_size - 2 * clip)
+    int begin_ss;                                           // number of first sub-span that is NOT completely clipped off
+    int end_ss;                                             // number of last sub-span that is NOT completely clipped off
+    int ss_bins[dMAX_STITCH];                               // number of bins delivered by eliminate()/Celiminate in each sub-span
+    volatile LONG input_busy[dMAX_STITCH][dMAX_NUM_FFT];
+    int num_pixels;                                         // number of pixels requested
+    int num_stitch;                                         // number of results to be stitched together to generate the pixel frame
+    unsigned long long stitch_flag;
+    int spec_flag[dMAX_STITCH];                             // flags showing if all ffts for a sub-span are done so elimination can proceed
+    double pix_per_bin;                                     // number of pixels per fft bin, note that this is fractional, not integral
+    double det_offset;                                      // offset needed in detector
+    double bin_per_pix;                                     // number of fft bins per pixel, this is fractional and != 1.0/pix_per_bin
+    double scale;                                           // output amplitude scale factor
+    double PiAlpha;                                         // parameter for Kaiser window function
+
+    int cal_set;                                            // specifies which set of calibration data to use
+    double f_min;                                           // frequency at first pixel (for calibration)
+    double f_max;                                           // frequency at last pixel (for calibration)
+    int cal_changed;                                        // flag to indicate that the calibration data has changed
+
+    double *window;                                         // pointer to buffer to hold window coefficients
+    double *result[dMAX_STITCH];                            // pointers to buffer to hold elimination results for each sub-span
+    dOUTREAL *pixels[dMAX_PIXOUTS][dNUM_PIXEL_BUFFS];       // pointers pixel output buffers
+    double *t_pixels[dMAX_PIXOUTS];                         // pointer to temporary pixel buffer                                    //pointer to temporary pixel buffer for non-averaged data
+    int w_pix_buff[dMAX_PIXOUTS];                           // number of pixel buffer owned by writing process
+    int r_pix_buff[dMAX_PIXOUTS];                           // number of pixel buffer owned by reading process
+    int last_pix_buff[dMAX_PIXOUTS];                        // number of the last pixel buffer written
+    volatile LONG pb_ready[dMAX_PIXOUTS][dNUM_PIXEL_BUFFS]; // if value is 0, this data has already been read; 1 = fresh data to read
+    int num_average[dMAX_PIXOUTS];                          // number of spans to average to create the pixels
+    int avail_frames[dMAX_PIXOUTS];                         // number of pixel frames currently available to average
+    int av_in_idx[dMAX_PIXOUTS];                            // input index in averaging pixel buffer ring
+    int av_out_idx[dMAX_PIXOUTS];                           // output index in averaging pixel buffer ring
+    double *av_sum[dMAX_PIXOUTS];                           // pointer to sum buffer for averaging
+    double *av_buff[dMAX_PIXOUTS][dMAX_AVERAGE];            // pointers to ring of buffers to hold pixel frames for averaging
+    double *pre_av_out;
+    int av_mode[dMAX_PIXOUTS];
+    double av_backmult[dMAX_PIXOUTS];                       // back multiplier for weighted averaging
+    double *cd;                                             // pointer to amplitude calibration buffer
+    int n_freqs[dMAX_CAL_SETS];                             // number of frequencies in each calibration set
+    double *freqs[dMAX_CAL_SETS];                           // pointers to vectors of calibration frequencies
+    double (*ac3[dMAX_CAL_SETS][dMAX_M]);                   // pointers to amplitude interpolant coefficients
+    double (*ac2[dMAX_CAL_SETS][dMAX_M]);
+    double (*ac1[dMAX_CAL_SETS][dMAX_M]);
+    double (*ac0[dMAX_CAL_SETS][dMAX_M]);
+
+    fftw_plan plan[dMAX_STITCH][dMAX_NUM_FFT];              // fftw plans
+    fftw_plan Cplan[dMAX_STITCH][dMAX_NUM_FFT];
+    double *fft_in[dMAX_STITCH][dMAX_NUM_FFT];              // pointers to fftw real input vectors
+    fftw_complex *Cfft_in[dMAX_STITCH][dMAX_NUM_FFT];       // pointers to fftw complex input vectors
+    fftw_complex *fft_out[dMAX_STITCH][dMAX_NUM_FFT];       // pointers to fftw complex output vectors
+    volatile LONG *pnum_threads;                            // pointer to current number of active worker threads
+    int stop;                                               // when set, fft threads will be returned to the pool
+    int end_dispatcher;                                     // set this flag to one to destroy the dispatcher thread
+    volatile int dispatcher;                                // one if the dispatcher thread is alive & active
+    int ss;                                                 // sub-span being processed
+    int LO;                                                 // LO (within current sub-span) being processed
+    int flag;
+    int have_samples[dMAX_STITCH][dMAX_NUM_FFT];            // number of unused samples remaining in a buffer
+    int type;                                               // 0 for REAL, 1 for COMPLEX
+    int incr;                                               // size - overlap
+    int buff_size;                                          // amount of data to be stored each time an input buffer is opened and closed = JanusAudio/BlockSize
+    dINREAL* I_samples[dMAX_STITCH][dMAX_NUM_FFT];          // pointers to current input position in I/Q buffers
+    dINREAL* Q_samples[dMAX_STITCH][dMAX_NUM_FFT];
+    int bsize;                                              // size of I_samples[][] and Q_samples[][] (number of samples they hold)
+    int IQout_index[dMAX_STITCH][dMAX_NUM_FFT];             // current output index for I_samples[ss][LO] and Q_samples[ss][LO]
+    int IQO_idx[dMAX_STITCH][dMAX_NUM_FFT];
+    int IQin_index[dMAX_STITCH][dMAX_NUM_FFT];              // current input index for I_samples[ss][LO] and Q_samples[ss][LO]
+    volatile LONG buff_ready[dMAX_STITCH][dMAX_NUM_FFT];    // 1 if buffer ready to read; 0 if needs to be filled
+    int max_writeahead;                                     // max allowed input samples ahead of where reading output samples
+
+    volatile LONG snap[dMAX_STITCH][dMAX_NUM_FFT];          // set to 1 to allow a snap of raw spectrum data
+    HANDLE hSnapEvent[dMAX_STITCH][dMAX_NUM_FFT];           // mutex handles; mutexes will be used to signal a snap is complete
+    double *snap_buff[dMAX_STITCH][dMAX_NUM_FFT];           // pointers to buffers for the snap
+
+    CRITICAL_SECTION PB_ControlsSection[dMAX_PIXOUTS];
+    CRITICAL_SECTION SetAnalyzerSection;
+    CRITICAL_SECTION BufferControlSection[dMAX_STITCH][dMAX_NUM_FFT];
+    CRITICAL_SECTION StitchSection;
+    CRITICAL_SECTION EliminateSection[dMAX_STITCH];
+    CRITICAL_SECTION ResampleSection;
+
+    int det_type[dMAX_PIXOUTS];                             // detector type
+    double inv_coherent_gain;
+    double inherent_power_gain;
+    double inv_enb;
+    double norm_oneHz;                                      // dB factor to normalize to one Hz bandwidth
+    int sample_rate;                                        // sample rate; used for normalization calculations
+    int normalize[dMAX_PIXOUTS];
+}  dp, *DP;
+
+extern DP pdisp[];
+
+extern __declspec( dllexport )
+void CreateAnalyzer (   int disp,
+                        int *success,
+                        char *app_data_path);
+
+extern __declspec( dllexport )
+void XCreateAnalyzer (  int disp,
+                        int *success,   //writes '0' to success if all went well, <0 if mem alloc failed
+                        int m_size,     //maximum fft size to be used
+                        int m_LO,       //maximum number of LO positions per subspan
+                        int m_stitch,   //maximum number of subspans to be concatenated
+                        char *app_data_path
+                     );
+
+extern __declspec( dllexport )
+void DestroyAnalyzer(int disp);
+
+extern __declspec( dllexport )
+void SetCalibration (   int disp,
+                        int set_num,                //identifier for this calibration data set
+                        int n_points,               //number of calibration points in the set
+                        double (*cal)[dMAX_M+1]     //pointer to the calibration table, first
+                    );
+
+extern __declspec( dllexport )
+void OpenBuffer(int disp, int ss, int LO, void **Ipointer, void **Qpointer);
+
+extern __declspec( dllexport )
+void CloseBuffer(int disp, int ss, int LO);
+
+extern __declspec( dllexport )
+void Spectrum(int disp, int ss, int LO, dINREAL* pI, dINREAL* pQ);
+
+extern __declspec( dllexport )
+void Spectrum2(int run, int disp, int ss, int LO, dINREAL* pbuff);
+
+extern __declspec( dllexport )
+void Spectrum0(int run, int disp, int ss, int LO, double* pbuff);
+
+extern __declspec( dllexport )
+void SnapSpectrum(  int disp,
+                    int ss,
+                    int LO,
+                    double *snap_buff);
+
+extern __declspec( dllexport )
+void SnapSpectrumTimeout (int disp,
+                          int ss,
+                          int LO,
+                          double* snap_buff,
+                          DWORD timeout,
+                          int* flag);
+
+#endif

wdsp/anf.c

@@ -0,0 +1,239 @@
+/*  anf.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"
+
+ANF create_anf  (
+                int run,
+                int position,
+                int buff_size,
+                double *in_buff,
+                double *out_buff,
+                int dline_size,
+                int n_taps,
+                int delay,
+                double two_mu,
+                double gamma,
+
+                double lidx,
+                double lidx_min,
+                double lidx_max,
+                double ngamma,
+                double den_mult,
+                double lincr,
+                double ldecr
+            )
+{
+    ANF a = (ANF) malloc0 (sizeof(anf));
+    a->run = run;
+    a->position = position;
+    a->buff_size = buff_size;
+    a->in_buff = in_buff;
+    a->out_buff = out_buff;
+    a->dline_size = dline_size;
+    a->mask = dline_size - 1;
+    a->n_taps = n_taps;
+    a->delay = delay;
+    a->two_mu = two_mu;
+    a->gamma = gamma;
+    a->in_idx = 0;
+    a->lidx = lidx;
+    a->lidx_min = lidx_min;
+    a->lidx_max = lidx_max;
+    a->ngamma = ngamma;
+    a->den_mult = den_mult;
+    a->lincr = lincr;
+    a->ldecr = ldecr;
+
+    memset (a->d, 0, sizeof(double) * ANF_DLINE_SIZE);
+    memset (a->w, 0, sizeof(double) * ANF_DLINE_SIZE);
+
+    return a;
+}
+
+void destroy_anf (ANF a)
+{
+    _aligned_free (a);
+}
+
+void xanf(ANF a, int position)
+{
+    int i, j, idx;
+    double c0, c1;
+    double y, error, sigma, inv_sigp;
+    double nel, nev;
+    if (a->run && (a->position == position))
+    {
+        for (i = 0; i < a->buff_size; i++)
+        {
+            a->d[a->in_idx] = a->in_buff[2 * i + 0];
+
+            y = 0;
+            sigma = 0;
+
+            for (j = 0; j < a->n_taps; j++)
+            {
+                idx = (a->in_idx + j + a->delay) & a->mask;
+                y += a->w[j] * a->d[idx];
+                sigma += a->d[idx] * a->d[idx];
+            }
+            inv_sigp = 1.0 / (sigma + 1e-10);
+            error = a->d[a->in_idx] - y;
+
+            a->out_buff[2 * i + 0] = error;
+            a->out_buff[2 * i + 1] = 0.0;
+
+            if((nel = error * (1.0 - a->two_mu * sigma * inv_sigp)) < 0.0) nel = -nel;
+            if((nev = a->d[a->in_idx] - (1.0 - a->two_mu * a->ngamma) * y - a->two_mu * error * sigma * inv_sigp) < 0.0) nev = -nev;
+            if (nev < nel)
+            {
+                if ((a->lidx += a->lincr) > a->lidx_max) a->lidx = a->lidx_max;
+            }
+            else
+            {
+                if ((a->lidx -= a->ldecr) < a->lidx_min) a->lidx = a->lidx_min;
+            }
+            a->ngamma = a->gamma * (a->lidx * a->lidx) * (a->lidx * a->lidx) * a->den_mult;
+
+            c0 = 1.0 - a->two_mu * a->ngamma;
+            c1 = a->two_mu * error * inv_sigp;
+
+            for (j = 0; j < a->n_taps; j++)
+            {
+                idx = (a->in_idx + j + a->delay) & a->mask;
+                a->w[j] = c0 * a->w[j] + c1 * a->d[idx];
+            }
+            a->in_idx = (a->in_idx + a->mask) & a->mask;
+        }
+    }
+    else if (a->in_buff != a->out_buff)
+        memcpy (a->out_buff, a->in_buff, a->buff_size * sizeof (complex));
+}
+
+void flush_anf (ANF a)
+{
+    memset (a->d, 0, sizeof(double) * ANF_DLINE_SIZE);
+    memset (a->w, 0, sizeof(double) * ANF_DLINE_SIZE);
+    a->in_idx = 0;
+}
+
+void setBuffers_anf (ANF a, double* in, double* out)
+{
+    a->in_buff = in;
+    a->out_buff = out;
+}
+
+void setSamplerate_anf (ANF a, int rate)
+{
+    flush_anf (a);
+}
+
+void setSize_anf (ANF a, int size)
+{
+    a->buff_size = size;
+    flush_anf (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetRXAANFRun (int channel, int run)
+{
+    ANF a = rxa[channel].anf.p;
+    if (a->run != run)
+    {
+        RXAbp1Check (channel, rxa[channel].amd.p->run, rxa[channel].snba.p->run,
+            rxa[channel].emnr.p->run, run, rxa[channel].anr.p->run);
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->run = run;
+        RXAbp1Set (channel);
+        flush_anf (a);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+
+PORT void
+SetRXAANFVals (int channel, int taps, int delay, double gain, double leakage)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->n_taps = taps;
+    rxa[channel].anf.p->delay = delay;
+    rxa[channel].anf.p->two_mu = gain;          //try two_mu = 1e-4
+    rxa[channel].anf.p->gamma = leakage;        //try gamma = 0.10
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANFTaps (int channel, int taps)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->n_taps = taps;
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANFDelay (int channel, int delay)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->delay = delay;
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANFGain (int channel, double gain)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->two_mu = gain;
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANFLeakage (int channel, double leakage)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->gamma = leakage;
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANFPosition (int channel, int position)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anf.p->position = position;
+    rxa[channel].bp1.p->position = position;
+    flush_anf (rxa[channel].anf.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/anf.h

@@ -0,0 +1,107 @@
+/*  anf.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 _anf_h
+#define _anf_h
+
+#define ANF_DLINE_SIZE 2048
+
+typedef struct _anf
+{
+    int run;
+    int position;
+    int buff_size;
+    double *in_buff;
+    double *out_buff;
+    int dline_size;
+    int mask;
+    int n_taps;
+    int delay;
+    double two_mu;
+    double gamma;
+    double d [ANF_DLINE_SIZE];
+    double w [ANF_DLINE_SIZE];
+    int in_idx;
+
+    double lidx;
+    double lidx_min;
+    double lidx_max;
+    double ngamma;
+    double den_mult;
+    double lincr;
+    double ldecr;
+} anf, *ANF;
+
+extern ANF create_anf   (
+                int run,
+                int position,
+                int buff_size,
+                double *in_buff,
+                double *out_buff,
+                int dline_size,
+                int n_taps,
+                int delay,
+                double two_mu,
+                double gamma,
+
+                double lidx,
+                double lidx_min,
+                double lidx_max,
+                double ngamma,
+                double den_mult,
+                double lincr,
+                double ldecr
+            );
+
+extern void destroy_anf (ANF a);
+
+extern void flush_anf (ANF a);
+
+extern void xanf (ANF a, int position);
+
+extern void setBuffers_anf (ANF a, double* in, double* out);
+
+extern void setSamplerate_anf (ANF a, int rate);
+
+extern void setSize_anf (ANF a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAANFRun (int channel, int setit);
+
+extern __declspec (dllexport) void SetRXAANFVals (int channel, int taps, int delay, double gain, double leakage);
+
+extern __declspec (dllexport) void SetRXAANFTaps (int channel, int taps);
+
+extern __declspec (dllexport) void SetRXAANFDelay (int channel, int delay);
+
+extern __declspec (dllexport) void SetRXAANFGain (int channel, double gain);
+
+extern __declspec (dllexport) void SetRXAANFLeakage (int channel, double leakage);
+
+extern __declspec (dllexport) void SetRXAANFPosition (int channel, int position);
+
+#endif

wdsp/anr.c

@@ -0,0 +1,238 @@
+/*  anr.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"
+
+ANR create_anr  (
+                int run,
+                int position,
+                int buff_size,
+                double *in_buff,
+                double *out_buff,
+                int dline_size,
+                int n_taps,
+                int delay,
+                double two_mu,
+                double gamma,
+
+                double lidx,
+                double lidx_min,
+                double lidx_max,
+                double ngamma,
+                double den_mult,
+                double lincr,
+                double ldecr
+            )
+{
+    ANR a = (ANR) malloc0 (sizeof(anr));
+    a->run = run;
+    a->position = position;
+    a->buff_size = buff_size;
+    a->in_buff = in_buff;
+    a->out_buff = out_buff;
+    a->dline_size = dline_size;
+    a->mask = dline_size - 1;
+    a->n_taps = n_taps;
+    a->delay = delay;
+    a->two_mu = two_mu;
+    a->gamma = gamma;
+    a->in_idx = 0;
+    a->lidx = lidx;
+    a->lidx_min = lidx_min;
+    a->lidx_max = lidx_max;
+    a->ngamma = ngamma;
+    a->den_mult = den_mult;
+    a->lincr = lincr;
+    a->ldecr = ldecr;
+
+    memset (a->d, 0, sizeof(double) * ANR_DLINE_SIZE);
+    memset (a->w, 0, sizeof(double) * ANR_DLINE_SIZE);
+
+    return a;
+}
+
+void destroy_anr (ANR a)
+{
+    _aligned_free (a);
+}
+
+void xanr (ANR a, int position)
+{
+    int i, j, idx;
+    double c0, c1;
+    double y, error, sigma, inv_sigp;
+    double nel, nev;
+    if (a->run && (a->position == position))
+    {
+        for (i = 0; i < a->buff_size; i++)
+        {
+            a->d[a->in_idx] = a->in_buff[2 * i + 0];
+
+            y = 0;
+            sigma = 0;
+
+            for (j = 0; j < a->n_taps; j++)
+            {
+                idx = (a->in_idx + j + a->delay) & a->mask;
+                y += a->w[j] * a->d[idx];
+                sigma += a->d[idx] * a->d[idx];
+            }
+            inv_sigp = 1.0 / (sigma + 1e-10);
+            error = a->d[a->in_idx] - y;
+
+            a->out_buff[2 * i + 0] = y;
+            a->out_buff[2 * i + 1] = 0.0;
+
+            if((nel = error * (1.0 - a->two_mu * sigma * inv_sigp)) < 0.0) nel = -nel;
+            if((nev = a->d[a->in_idx] - (1.0 - a->two_mu * a->ngamma) * y - a->two_mu * error * sigma * inv_sigp) < 0.0) nev = -nev;
+            if (nev < nel)
+            {
+                if ((a->lidx += a->lincr) > a->lidx_max) a->lidx = a->lidx_max;
+            }
+            else
+            {
+                if ((a->lidx -= a->ldecr) < a->lidx_min) a->lidx = a->lidx_min;
+            }
+            a->ngamma = a->gamma * (a->lidx * a->lidx) * (a->lidx * a->lidx) * a->den_mult;
+
+            c0 = 1.0 - a->two_mu * a->ngamma;
+            c1 = a->two_mu * error * inv_sigp;
+
+            for (j = 0; j < a->n_taps; j++)
+            {
+                idx = (a->in_idx + j + a->delay) & a->mask;
+                a->w[j] = c0 * a->w[j] + c1 * a->d[idx];
+            }
+            a->in_idx = (a->in_idx + a->mask) & a->mask;
+        }
+    }
+    else if (a->in_buff != a->out_buff)
+        memcpy (a->out_buff, a->in_buff, a->buff_size * sizeof (complex));
+}
+
+void flush_anr (ANR a)
+{
+    memset (a->d, 0, sizeof(double) * ANR_DLINE_SIZE);
+    memset (a->w, 0, sizeof(double) * ANR_DLINE_SIZE);
+    a->in_idx = 0;
+}
+
+void setBuffers_anr (ANR a, double* in, double* out)
+{
+    a->in_buff = in;
+    a->out_buff = out;
+}
+
+void setSamplerate_anr (ANR a, int rate)
+{
+    flush_anr(a);
+}
+
+void setSize_anr (ANR a, int size)
+{
+    a->buff_size = size;
+    flush_anr(a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetRXAANRRun (int channel, int run)
+{
+    ANR a = rxa[channel].anr.p;
+    if (a->run != run)
+    {
+        RXAbp1Check (channel, rxa[channel].amd.p->run, rxa[channel].snba.p->run,
+            rxa[channel].emnr.p->run, rxa[channel].anf.p->run, run);
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->run = run;
+        RXAbp1Set (channel);
+        flush_anr (a);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT void
+SetRXAANRVals (int channel, int taps, int delay, double gain, double leakage)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->n_taps = taps;
+    rxa[channel].anr.p->delay = delay;
+    rxa[channel].anr.p->two_mu = gain;
+    rxa[channel].anr.p->gamma = leakage;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANRTaps (int channel, int taps)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->n_taps = taps;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANRDelay (int channel, int delay)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->delay = delay;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANRGain (int channel, double gain)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->two_mu = gain;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANRLeakage (int channel, double leakage)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->gamma = leakage;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT void
+SetRXAANRPosition (int channel, int position)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].anr.p->position = position;
+    rxa[channel].bp1.p->position = position;
+    flush_anr (rxa[channel].anr.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/anr.h

@@ -0,0 +1,107 @@
+/*  anr.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 _anr_h
+#define _anr_h
+
+#define ANR_DLINE_SIZE 2048
+
+typedef struct _anr
+{
+    int run;
+    int position;
+    int buff_size;
+    double *in_buff;
+    double *out_buff;
+    int dline_size;
+    int mask;
+    int n_taps;
+    int delay;
+    double two_mu;
+    double gamma;
+    double d [ANR_DLINE_SIZE];
+    double w [ANR_DLINE_SIZE];
+    int in_idx;
+
+    double lidx;
+    double lidx_min;
+    double lidx_max;
+    double ngamma;
+    double den_mult;
+    double lincr;
+    double ldecr;
+} anr, *ANR;
+
+extern ANR create_anr   (
+                int run,
+                int position,
+                int buff_size,
+                double *in_buff,
+                double *out_buff,
+                int dline_size,
+                int n_taps,
+                int delay,
+                double two_mu,
+                double gamma,
+
+                double lidx,
+                double lidx_min,
+                double lidx_max,
+                double ngamma,
+                double den_mult,
+                double lincr,
+                double ldecr
+            );
+
+extern void destroy_anr (ANR a);
+
+extern void flush_anr (ANR a);
+
+extern void xanr (ANR a, int position);
+
+extern void setBuffers_anr (ANR a, double* in, double* out);
+
+extern void setSamplerate_anr (ANR a, int rate);
+
+extern void setSize_anr (ANR a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAANRRun (int channel, int setit);
+
+extern __declspec (dllexport) void SetRXAANRVals (int channel, int taps, int delay, double gain, double leakage);
+
+extern __declspec (dllexport) void SetRXAANRTaps (int channel, int taps);
+
+extern __declspec (dllexport) void SetRXAANRDelay (int channel, int delay);
+
+extern __declspec (dllexport) void SetRXAANRGain (int channel, double gain);
+
+extern __declspec (dllexport) void SetRXAANRLeakage (int channel, double leakage);
+
+extern __declspec (dllexport) void SetRXAANRPosition (int channel, int position);
+
+#endif

wdsp/bandpass.c

@@ -0,0 +1,594 @@
+/*  bandpass.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2016, 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"
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       Overlap-Save Bandpass                                           *
+*                                                                                                       *
+********************************************************************************************************/
+
+void calc_bps (BPS a)
+{
+    double* impulse;
+    a->infilt = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->product = (double *)malloc0(2 * a->size * sizeof(complex));
+    impulse = fir_bandpass(a->size + 1, a->f_low, a->f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+    a->mults = fftcv_mults(2 * a->size, impulse);
+    a->CFor = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->infilt, (fftw_complex *)a->product, FFTW_FORWARD, FFTW_PATIENT);
+    a->CRev = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->product, (fftw_complex *)a->out, FFTW_BACKWARD, FFTW_PATIENT);
+    _aligned_free(impulse);
+}
+
+void decalc_bps (BPS a)
+{
+    fftw_destroy_plan(a->CRev);
+    fftw_destroy_plan(a->CFor);
+    _aligned_free(a->mults);
+    _aligned_free(a->product);
+    _aligned_free(a->infilt);
+}
+
+BPS create_bps (int run, int position, int size, double* in, double* out,
+    double f_low, double f_high, int samplerate, int wintype, double gain)
+{
+    BPS a = (BPS) malloc0 (sizeof (bps));
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->samplerate = (double)samplerate;
+    a->wintype = wintype;
+    a->gain = gain;
+    a->in = in;
+    a->out = out;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    calc_bps (a);
+    return a;
+}
+
+void destroy_bps (BPS a)
+{
+    decalc_bps (a);
+    _aligned_free (a);
+}
+
+void flush_bps (BPS a)
+{
+    memset (a->infilt, 0, 2 * a->size * sizeof (complex));
+}
+
+void xbps (BPS a, int pos)
+{
+    int i;
+    double I, Q;
+    if (a->run && pos == a->position)
+    {
+        memcpy (&(a->infilt[2 * a->size]), a->in, a->size * sizeof (complex));
+        fftw_execute (a->CFor);
+        for (i = 0; i < 2 * a->size; i++)
+        {
+            I = a->gain * a->product[2 * i + 0];
+            Q = a->gain * a->product[2 * i + 1];
+            a->product[2 * i + 0] = I * a->mults[2 * i + 0] - Q * a->mults[2 * i + 1];
+            a->product[2 * i + 1] = I * a->mults[2 * i + 1] + Q * a->mults[2 * i + 0];
+        }
+        fftw_execute (a->CRev);
+        memcpy (a->infilt, &(a->infilt[2 * a->size]), a->size * sizeof(complex));
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_bps (BPS a, double* in, double* out)
+{
+    decalc_bps (a);
+    a->in = in;
+    a->out = out;
+    calc_bps (a);
+}
+
+void setSamplerate_bps (BPS a, int rate)
+{
+    decalc_bps (a);
+    a->samplerate = rate;
+    calc_bps (a);
+}
+
+void setSize_bps (BPS a, int size)
+{
+    decalc_bps (a);
+    a->size = size;
+    calc_bps (a);
+}
+
+void setFreqs_bps (BPS a, double f_low, double f_high)
+{
+    decalc_bps (a);
+    a->f_low = f_low;
+    a->f_high = f_high;
+    calc_bps (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Overlap-Save Bandpass:  RXA Properties                                  *
+*                                                                                                       *
+********************************************************************************************************/
+/*  // UNCOMMENT properties when a pointer is in place in rxa[channel]
+PORT
+void SetRXABPSRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].bp1.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXABPSFreqs (int channel, double f_low, double f_high)
+{
+    double* impulse;
+    BPS a1;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a1 = rxa[channel].bp1.p;
+    if ((f_low != a1->f_low) || (f_high != a1->f_high))
+    {
+        a1->f_low = f_low;
+        a1->f_high = f_high;
+        _aligned_free (a1->mults);
+        impulse = fir_bandpass(a1->size + 1, f_low, f_high, a1->samplerate, a1->wintype, 1, 1.0 / (double)(2 * a1->size));
+        a1->mults = fftcv_mults (2 * a1->size, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXABPSWindow (int channel, int wintype)
+{
+    double* impulse;
+    BPS a1;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a1 = rxa[channel].bp1.p;
+    if ((a1->wintype != wintype))
+    {
+        a1->wintype = wintype;
+        _aligned_free (a1->mults);
+        impulse = fir_bandpass(a1->size + 1, a1->f_low, a1->f_high, a1->samplerate, a1->wintype, 1, 1.0 / (double)(2 * a1->size));
+        a1->mults = fftcv_mults (2 * a1->size, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+*/
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+/*  // UNCOMMENT properties when pointers in place in txa[channel]
+PORT
+void SetTXABPSRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].bp1.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXABPSFreqs (int channel, double f_low, double f_high)
+{
+    double* impulse;
+    BPS a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].bp0.p;
+    if ((f_low != a->f_low) || (f_high != a->f_high))
+    {
+        a->f_low = f_low;
+        a->f_high = f_high;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass(a->size + 1, f_low, f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp1.p;
+    if ((f_low != a->f_low) || (f_high != a->f_high))
+    {
+        a->f_low = f_low;
+        a->f_high = f_high;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass(a->size + 1, f_low, f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp2.p;
+    if ((f_low != a->f_low) || (f_high != a->f_high))
+    {
+        a->f_low = f_low;
+        a->f_high = f_high;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass(a->size + 1, f_low, f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXABPSWindow (int channel, int wintype)
+{
+    double* impulse;
+    BPS a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].bp0.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass(a->size + 1, a->f_low, a->f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp1.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass(a->size + 1, a->f_low, a->f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp2.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        _aligned_free (a->mults);
+        impulse = fir_bandpass (a->size + 1, a->f_low, a->f_high, a->samplerate, a->wintype, 1, 1.0 / (double)(2 * a->size));
+        a->mults = fftcv_mults (2 * a->size, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+*/
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Bandpass                                   *
+*                                                                                                       *
+********************************************************************************************************/
+
+BANDPASS create_bandpass (int run, int position, int size, int nc, int mp, double* in, double* out,
+    double f_low, double f_high, int samplerate, int wintype, double gain)
+{
+    // NOTE:  'nc' must be >= 'size'
+    BANDPASS a = (BANDPASS) malloc0 (sizeof (bandpass));
+    double* impulse;
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->nc = nc;
+    a->mp = mp;
+    a->in = in;
+    a->out = out;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    a->samplerate = samplerate;
+    a->wintype = wintype;
+    a->gain = gain;
+    impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+    a->p = create_fircore (a->size, a->in, a->out, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+    return a;
+}
+
+void destroy_bandpass (BANDPASS a)
+{
+    destroy_fircore (a->p);
+    _aligned_free (a);
+}
+
+void flush_bandpass (BANDPASS a)
+{
+    flush_fircore (a->p);
+}
+
+void xbandpass (BANDPASS a, int pos)
+{
+    if (a->run && a->position == pos)
+        xfircore (a->p);
+    else if (a->out != a->in)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_bandpass (BANDPASS a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    setBuffers_fircore (a->p, a->in, a->out);
+}
+
+void setSamplerate_bandpass (BANDPASS a, int rate)
+{
+    double* impulse;
+    a->samplerate = rate;
+    impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+void setSize_bandpass (BANDPASS a, int size)
+{
+    // NOTE:  'size' must be <= 'nc'
+    double* impulse;
+    a->size = size;
+    setSize_fircore (a->p, a->size);
+    // recalc impulse because scale factor is a function of size
+    impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+void setGain_bandpass (BANDPASS a, double gain, int update)
+{
+    double* impulse;
+    a->gain = gain;
+    impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+    setImpulse_fircore (a->p, impulse, update);
+    _aligned_free (impulse);
+}
+
+void CalcBandpassFilter (BANDPASS a, double f_low, double f_high, double gain)
+{
+    double* impulse;
+    if ((a->f_low != f_low) || (a->f_high != f_high) || (a->gain != gain))
+    {
+        a->f_low = f_low;
+        a->f_high = f_high;
+        a->gain = gain;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXABandpassRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].bp1.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXABandpassFreqs (int channel, double f_low, double f_high)
+{
+    double* impulse;
+    BANDPASS a = rxa[channel].bp1.p;
+    if ((f_low != a->f_low) || (f_high != a->f_high))
+    {
+        impulse = fir_bandpass (a->nc, f_low, f_high, a->samplerate,
+            a->wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 0);
+        _aligned_free (impulse);
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->f_low = f_low;
+        a->f_high = f_high;
+        setUpdate_fircore (a->p);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetRXABandpassWindow (int channel, int wintype)
+{
+    double* impulse;
+    BANDPASS a = rxa[channel].bp1.p;
+    if ((a->wintype != wintype))
+    {
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate,
+            wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 0);
+        _aligned_free (impulse);
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->wintype = wintype;
+        setUpdate_fircore (a->p);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetRXABandpassNC (int channel, int nc)
+{
+    // NOTE:  'nc' must be >= 'size'
+    double* impulse;
+    BANDPASS a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].bp1.p;
+    if (nc != a->nc)
+    {
+        a->nc = nc;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXABandpassMP (int channel, int mp)
+{
+    BANDPASS a;
+    a = rxa[channel].bp1.p;
+    if (mp != a->mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXABandpassRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].bp1.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+//PORT
+//void SetTXABandpassFreqs (int channel, double f_low, double f_high)
+//{
+//  double* impulse;
+//  BANDPASS a;
+//  a = txa[channel].bp0.p;
+//  if ((f_low != a->f_low) || (f_high != a->f_high))
+//  {
+//      a->f_low = f_low;
+//      a->f_high = f_high;
+//      impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+//      setImpulse_fircore (a->p, impulse, 1);
+//      _aligned_free (impulse);
+//  }
+//  a = txa[channel].bp1.p;
+//  if ((f_low != a->f_low) || (f_high != a->f_high))
+//  {
+//      a->f_low = f_low;
+//      a->f_high = f_high;
+//      impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+//      setImpulse_fircore (a->p, impulse, 1);
+//      _aligned_free (impulse);
+//  }
+//  a = txa[channel].bp2.p;
+//  if ((f_low != a->f_low) || (f_high != a->f_high))
+//  {
+//      a->f_low = f_low;
+//      a->f_high = f_high;
+//      impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+//      setImpulse_fircore (a->p, impulse, 1);
+//      _aligned_free (impulse);
+//  }
+//}
+
+PORT
+void SetTXABandpassWindow (int channel, int wintype)
+{
+    double* impulse;
+    BANDPASS a;
+    a = txa[channel].bp0.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp1.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp2.p;
+    if (a->wintype != wintype)
+    {
+        a->wintype = wintype;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+}
+
+PORT
+void SetTXABandpassNC (int channel, int nc)
+{
+    // NOTE:  'nc' must be >= 'size'
+    double* impulse;
+    BANDPASS a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].bp0.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp1.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    a = txa[channel].bp2.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain / (double)(2 * a->size));
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXABandpassMP (int channel, int mp)
+{
+    BANDPASS a;
+    a = txa[channel].bp0.p;
+    if (mp != a->mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+    a = txa[channel].bp1.p;
+    if (mp != a->mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+    a = txa[channel].bp2.p;
+    if (mp != a->mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}

wdsp/bandpass.h

@@ -0,0 +1,143 @@
+/*  bandpass.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2016, 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
+
+*/
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       Overlap-Save Bandpass                                           *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _bps_h
+#define _bps_h
+
+typedef struct _bps
+{
+    int run;
+    int position;
+    int size;
+    double* in;
+    double* out;
+    double f_low;
+    double f_high;
+    double* infilt;
+    double* product;
+    double* mults;
+    double samplerate;
+    int wintype;
+    double gain;
+    fftw_plan CFor;
+    fftw_plan CRev;
+}bps, *BPS;
+
+extern BPS create_bps (int run, int position, int size, double* in, double* out,
+    double f_low, double f_high, int samplerate, int wintype, double gain);
+
+extern void destroy_bps (BPS a);
+
+extern void flush_bps (BPS a);
+
+extern void xbps (BPS a, int pos);
+
+extern void setBuffers_bps (BPS a, double* in, double* out);
+
+extern void setSamplerate_bps (BPS a, int rate);
+
+extern void setSize_bps (BPS a, int size);
+
+extern void setFreqs_bps (BPS a, double f_low, double f_high);
+
+// RXA Prototypes
+
+extern __declspec (dllexport) void SetRXABPSRun (int channel, int run);
+
+extern __declspec (dllexport) void SetRXABPSFreqs (int channel, double low, double high);
+
+// TXA Prototypes
+
+extern __declspec (dllexport) void SetTXABPSRun (int channel, int run);
+
+extern __declspec (dllexport) void SetTXABPSFreqs (int channel, double low, double high);
+
+#endif
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Bandpass                                   *
+*                                                                                                       *
+********************************************************************************************************/
+
+
+#ifndef _bandpass_h
+#define _bandpass_h
+#include "firmin.h"
+typedef struct _bandpass
+{
+    int run;
+    int position;
+    int size;
+    int nc;
+    int mp;
+    double* in;
+    double* out;
+    double f_low;
+    double f_high;
+    double samplerate;
+    int wintype;
+    double gain;
+    FIRCORE p;
+}bandpass, *BANDPASS;
+
+extern BANDPASS create_bandpass (int run, int position, int size, int nc, int mp, double* in, double* out,
+    double f_low, double f_high, int samplerate, int wintype, double gain);
+
+extern void destroy_bandpass (BANDPASS a);
+
+extern void flush_bandpass (BANDPASS a);
+
+extern void xbandpass (BANDPASS a, int pos);
+
+extern void setBuffers_bandpass (BANDPASS a, double* in, double* out);
+
+extern void setSamplerate_bandpass (BANDPASS a, int rate);
+
+extern void setSize_bandpass (BANDPASS a, int size);
+
+extern void setGain_bandpass (BANDPASS a, double gain, int update);
+
+extern void CalcBandpassFilter (BANDPASS a, double f_low, double f_high, double gain);
+
+extern __declspec (dllexport) void SetRXABandpassFreqs (int channel, double f_low, double f_high);
+
+extern __declspec (dllexport) void SetRXABandpassNC (int channel, int nc);
+
+extern __declspec (dllexport) void SetRXABandpassMP (int channel, int mp);
+
+extern __declspec (dllexport) void SetTXABandpassNC (int channel, int nc);
+
+extern __declspec (dllexport) void SetTXABandpassMP (int channel, int mp);
+
+#endif

wdsp/calcc.h

@@ -0,0 +1,167 @@
+/*  calcc.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 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
+
+*/
+
+#ifndef _calcc_h
+#define _calcc_h
+#include "delay.h"
+#include "lmath.h"
+typedef struct _calcc
+{
+    int channel;
+    int runcal;
+    int size;
+    volatile long mox;
+    volatile long solidmox;
+    int rate;
+    int ints;
+    int spi;
+    int nsamps;
+    int npsamps;
+    int pin;
+    int map;
+    int convex;
+    int stbl;
+    int scOK;
+    double hw_scale;
+    double rx_scale;
+    double alpha;
+
+    int tsamps;
+    double* env_TX;
+    double* env_RX;
+    double* x;
+    double* ym;
+    double* yc;
+    double* ys;
+    double* cat;
+
+    double* t;
+    double* tmap;
+    double* cm;
+    double* cc;
+    double* cs;
+    double* cm_old;
+    double* rxs;
+    double* txs;
+    double ptol;
+    int* info;
+    int* binfo;
+    double txdel;
+    BLDR ccbld;
+    volatile long savecorr_bypass;
+    HANDLE Sem_SaveCorr;
+    volatile long restcorr_bypass;
+    HANDLE Sem_RestCorr;
+    volatile long calccorr_bypass;
+    HANDLE Sem_CalcCorr;
+    volatile long turnoff_bypass;
+    HANDLE Sem_TurnOff;
+    struct _ctrl
+    {
+        double moxdelay;
+        double loopdelay;
+        int state;
+        int reset;
+        int automode;
+        int mancal;
+        int turnon;
+        int moxsamps;
+        int moxcount;
+        int count;
+        int* cpi;
+        int* sindex;
+        int* sbase;
+        int full_ints;
+        int calcinprogress;
+        volatile LONG calcdone;
+        int waitsamps;
+        int waitcount;
+        double env_maxtx;
+        volatile long running;
+        int bs_count;
+        volatile long current_state;
+        CRITICAL_SECTION cs_SafeToEnd;
+    } ctrl;
+    struct _disp
+    {
+        double* x;
+        double* ym;
+        double* yc;
+        double* ys;
+        double* cm;
+        double* cc;
+        double* cs;
+        CRITICAL_SECTION cs_disp;
+    } disp;
+    DELAY rxdelay;
+    DELAY txdelay;
+    struct _util
+    {
+        char savefile[256];
+        char restfile[256];
+        int ints;
+        int channel;
+        double* pm;
+        double* pc;
+        double* ps;
+    } util;
+    double* temptx;             //////////////////////////////////////////////////// temporary tx complex buffer - remove with new callback3port()
+    double* temprx;             //////////////////////////////////////////////////// temporary rx complex buffer - remove with new callback3port()
+} calcc, *CALCC;
+
+extern CALCC create_calcc (int channel, int runcal, int size, int rate, int ints, int spi, double hw_scale,
+    double moxdelay, double loopdelay, double ptol, int mox, int solidmox, int pin, int map, int stbl,
+    int npsamps, double alpha);
+
+extern void destroy_calcc (CALCC a);
+
+extern void flush_calcc (CALCC a);
+
+extern __declspec(dllexport) void pscc (int channel, int size, double* tx, double* rx);
+
+extern void __cdecl PSSaveCorrection(void* pargs);
+
+extern void __cdecl PSRestoreCorrection(void* pargs);
+
+extern void __cdecl doPSCalcCorrection(void* arg);
+
+extern void __cdecl doPSTurnoff(void* arg);
+
+#endif
+
+// 'info' assignments:
+//       0 - builder for rx_scale
+//       1 - builder for cm
+//       2 - builder for cc
+//       3 - builder for cs
+//       4 - feedback level warning
+//       5 - count of attempted calibrations
+//       6 - results from scheck()
+//       7 - results from rxscheck()
+//
+//      13 - dogcount
+//      14 - indicates iqc_Run = 1
+//      15 - control state

wdsp/calculus.h

@@ -0,0 +1,8 @@
+#ifndef _calculus_h
+#define _calculus_h
+
+extern double GG[];
+
+extern double GGS[];
+
+#endif

wdsp/cblock.c

@@ -0,0 +1,126 @@
+/*  cblock.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 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
+
+*/
+
+#include "comm.h"
+
+void calc_cbl (CBL a)
+{
+    a->prevIin  = 0.0;
+    a->prevQin  = 0.0;
+    a->prevIout = 0.0;
+    a->prevQout = 0.0;
+    a->mtau = exp(-1.0 / (a->sample_rate * a->tau));
+}
+
+CBL create_cbl
+    (
+    int run,
+    int buff_size,
+    double *in_buff,
+    double *out_buff,
+    int mode,
+    int sample_rate,
+    double tau
+    )
+{
+    CBL a = (CBL) malloc0 (sizeof(cbl));
+    a->run = run;
+    a->buff_size = buff_size;
+    a->in_buff = in_buff;
+    a->out_buff = out_buff;
+    a->mode = mode;
+    a->sample_rate = (double)sample_rate;
+    a->tau = tau;
+    calc_cbl (a);
+    return a;
+}
+
+void destroy_cbl(CBL a)
+{
+    _aligned_free (a);
+}
+
+void flush_cbl (CBL a)
+{
+    a->prevIin  = 0.0;
+    a->prevQin  = 0.0;
+    a->prevIout = 0.0;
+    a->prevQout = 0.0;
+}
+
+void xcbl (CBL a)
+{
+    if (a->run)
+    {
+        int i;
+        double tempI, tempQ;
+        for (i = 0; i < a->buff_size; i++)
+        {
+            tempI  = a->in_buff[2 * i + 0];
+            tempQ  = a->in_buff[2 * i + 1];
+            a->out_buff[2 * i + 0] = a->in_buff[2 * i + 0] - a->prevIin + a->mtau * a->prevIout;
+            a->out_buff[2 * i + 1] = a->in_buff[2 * i + 1] - a->prevQin + a->mtau * a->prevQout;
+            a->prevIin  = tempI;
+            a->prevQin  = tempQ;
+            if (fabs(a->prevIout = a->out_buff[2 * i + 0]) < 1.0e-100) a->prevIout = 0.0;
+            if (fabs(a->prevQout = a->out_buff[2 * i + 1]) < 1.0e-100) a->prevQout = 0.0;
+        }
+    }
+    else if (a->in_buff != a->out_buff)
+        memcpy (a->out_buff, a->in_buff, a->buff_size * sizeof (complex));
+}
+
+void setBuffers_cbl (CBL a, double* in, double* out)
+{
+    a->in_buff = in;
+    a->out_buff = out;
+}
+
+void setSamplerate_cbl (CBL a, int rate)
+{
+    a->sample_rate = rate;
+    calc_cbl (a);
+}
+
+void setSize_cbl (CBL a, int size)
+{
+    a->buff_size = size;
+    flush_cbl (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetRXACBLRun(int channel, int setit)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].cbl.p->run = setit;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/cblock.h

@@ -0,0 +1,73 @@
+/*  cblock.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 _cblock_h
+#define _cblock_h
+
+typedef struct _cbl
+{
+    int run;                            //run
+    int buff_size;                      //buffer size
+    double *in_buff;                    //pointer to input buffer
+    double *out_buff;                   //pointer to output buffer
+    int mode;
+    double sample_rate;                 //sample rate
+    double prevIin;
+    double prevQin;
+    double prevIout;
+    double prevQout;
+    double tau;                         //carrier removal time constant
+    double mtau;                        //carrier removal multiplier
+} cbl, *CBL;
+
+extern CBL create_cbl
+    (
+    int run,
+    int buff_size,
+    double *in_buff,
+    double *out_buff,
+    int mode,
+    int sample_rate,
+    double tau
+    );
+
+extern void destroy_cbl (CBL a);
+
+extern void flush_cbl (CBL a);
+
+extern void xcbl (CBL a);
+
+extern void setBuffers_cbl (CBL a, double* in, double* out);
+
+extern void setSamplerate_cbl (CBL a, int rate);
+
+extern void setSize_cbl (CBL a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXACBLRun(int channel, int setit);
+
+#endif

wdsp/cfcomp.c

@@ -0,0 +1,519 @@
+/*  cfcomp.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2017, 2021 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 calc_cfcwindow (CFCOMP a)
+{
+    int i;
+    double arg0, arg1, cgsum, igsum, coherent_gain, inherent_power_gain, wmult;
+    switch (a->wintype)
+    {
+    case 0:
+        arg0 = 2.0 * PI / (double)a->fsize;
+        cgsum = 0.0;
+        igsum = 0.0;
+        for (i = 0; i < a->fsize; i++)
+        {
+            a->window[i] = sqrt (0.54 - 0.46 * cos((double)i * arg0));
+            cgsum += a->window[i];
+            igsum += a->window[i] * a->window[i];
+        }
+        coherent_gain = cgsum / (double)a->fsize;
+        inherent_power_gain = igsum / (double)a->fsize;
+        wmult = 1.0 / sqrt (inherent_power_gain);
+        for (i = 0; i < a->fsize; i++)
+            a->window[i] *= wmult;
+        a->winfudge = sqrt (1.0 / coherent_gain);
+        break;
+    case 1:
+        arg0 = 2.0 * PI / (double)a->fsize;
+        cgsum = 0.0;
+        igsum = 0.0;
+        for (i = 0; i < a->fsize; i++)
+        {
+            arg1 = cos(arg0 * (double)i);
+            a->window[i]  = sqrt   (+0.21747
+                          + arg1 * (-0.45325
+                          + arg1 * (+0.28256
+                          + arg1 * (-0.04672))));
+            cgsum += a->window[i];
+            igsum += a->window[i] * a->window[i];
+        }
+        coherent_gain = cgsum / (double)a->fsize;
+        inherent_power_gain = igsum / (double)a->fsize;
+        wmult = 1.0 / sqrt (inherent_power_gain);
+        for (i = 0; i < a->fsize; i++)
+            a->window[i] *= wmult;
+        a->winfudge = sqrt (1.0 / coherent_gain);
+        break;
+    }
+}
+
+int fCOMPcompare (const void * a, const void * b)
+{
+    if (*(double*)a < *(double*)b)
+        return -1;
+    else if (*(double*)a == *(double*)b)
+        return 0;
+    else
+        return 1;
+}
+
+void calc_comp (CFCOMP a)
+{
+    int i, j;
+    double f, frac, fincr, fmax;
+    double* sary;
+    a->precomplin = pow (10.0, 0.05 * a->precomp);
+    a->prepeqlin  = pow (10.0, 0.05 * a->prepeq);
+    fmax = 0.5 * a->rate;
+    for (i = 0; i < a->nfreqs; i++)
+    {
+        a->F[i] = max (a->F[i], 0.0);
+        a->F[i] = min (a->F[i], fmax);
+        a->G[i] = max (a->G[i], 0.0);
+    }
+    sary = (double *)malloc0 (3 * a->nfreqs * sizeof (double));
+    for (i = 0; i < a->nfreqs; i++)
+    {
+        sary[3 * i + 0] = a->F[i];
+        sary[3 * i + 1] = a->G[i];
+        sary[3 * i + 2] = a->E[i];
+    }
+    qsort (sary, a->nfreqs, 3 * sizeof (double), fCOMPcompare);
+    for (i = 0; i < a->nfreqs; i++)
+    {
+        a->F[i] = sary[3 * i + 0];
+        a->G[i] = sary[3 * i + 1];
+        a->E[i] = sary[3 * i + 2];
+    }
+    _aligned_free (sary);
+    a->fp[0] = 0.0;
+    a->fp[a->nfreqs + 1] = fmax;
+    a->gp[0] = a->G[0];
+    a->gp[a->nfreqs + 1] = a->G[a->nfreqs - 1];
+    a->ep[0] = a->E[0];                             // cutoff?
+    a->ep[a->nfreqs + 1] = a->E[a->nfreqs - 1];     // cutoff?
+    for (i = 0, j = 1; i < a->nfreqs; i++, j++)
+    {
+        a->fp[j] = a->F[i];
+        a->gp[j] = a->G[i];
+        a->ep[j] = a->E[i];
+    }
+    fincr = a->rate / (double)a->fsize;
+    j = 0;
+    // print_impulse ("gp.txt", a->nfreqs+2, a->gp, 0, 0);
+    for (i = 0; i < a->msize; i++)
+    {
+        f = fincr * (double)i;
+        while (f >= a->fp[j + 1] && j < a->nfreqs) j++;
+        frac = (f - a->fp[j]) / (a->fp[j + 1] - a->fp[j]);
+        a->comp[i] = pow (10.0, 0.05 * (frac * a->gp[j + 1] + (1.0 - frac) * a->gp[j]));
+        a->peq[i]  = pow (10.0, 0.05 * (frac * a->ep[j + 1] + (1.0 - frac) * a->ep[j]));
+        a->cfc_gain[i] = a->precomplin * a->comp[i];
+    }
+    // print_impulse ("comp.txt", a->msize, a->comp, 0, 0);
+}
+
+void calc_cfcomp(CFCOMP a)
+{
+    int i;
+    a->incr = a->fsize / a->ovrlp;
+    if (a->fsize > a->bsize)
+        a->iasize = a->fsize;
+    else
+        a->iasize = a->bsize + a->fsize - a->incr;
+    a->iainidx = 0;
+    a->iaoutidx = 0;
+    if (a->fsize > a->bsize)
+    {
+        if (a->bsize > a->incr)  a->oasize = a->bsize;
+        else                     a->oasize = a->incr;
+        a->oainidx = (a->fsize - a->bsize - a->incr) % a->oasize;
+    }
+    else
+    {
+        a->oasize = a->bsize;
+        a->oainidx = a->fsize - a->incr;
+    }
+    a->init_oainidx = a->oainidx;
+    a->oaoutidx = 0;
+    a->msize = a->fsize / 2 + 1;
+    a->window    = (double *)malloc0 (a->fsize  * sizeof(double));
+    a->inaccum   = (double *)malloc0 (a->iasize * sizeof(double));
+    a->forfftin  = (double *)malloc0 (a->fsize  * sizeof(double));
+    a->forfftout = (double *)malloc0 (a->msize  * sizeof(complex));
+    a->cmask     = (double *)malloc0 (a->msize  * sizeof(double));
+    a->mask      = (double *)malloc0 (a->msize  * sizeof(double));
+    a->cfc_gain  = (double *)malloc0 (a->msize  * sizeof(double));
+    a->revfftin  = (double *)malloc0 (a->msize  * sizeof(complex));
+    a->revfftout = (double *)malloc0 (a->fsize  * sizeof(double));
+    a->save      = (double **)malloc0(a->ovrlp  * sizeof(double *));
+    for (i = 0; i < a->ovrlp; i++)
+        a->save[i] = (double *)malloc0(a->fsize * sizeof(double));
+    a->outaccum = (double *)malloc0(a->oasize * sizeof(double));
+    a->nsamps = 0;
+    a->saveidx = 0;
+    a->Rfor = fftw_plan_dft_r2c_1d(a->fsize, a->forfftin, (fftw_complex *)a->forfftout, FFTW_ESTIMATE);
+    a->Rrev = fftw_plan_dft_c2r_1d(a->fsize, (fftw_complex *)a->revfftin, a->revfftout, FFTW_ESTIMATE);
+    calc_cfcwindow(a);
+
+    a->pregain  = (2.0 * a->winfudge) / (double)a->fsize;
+    a->postgain = 0.5 / ((double)a->ovrlp * a->winfudge);
+
+    a->fp = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    a->gp = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    a->ep = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    a->comp = (double *) malloc0 (a->msize * sizeof (double));
+    a->peq  = (double *) malloc0 (a->msize * sizeof (double));
+    calc_comp (a);
+
+    a->gain = 0.0;
+    a->mmult = exp (-1.0 / (a->rate * a->ovrlp * a->mtau));
+    a->dmult = exp (-(double)a->fsize / (a->rate * a->ovrlp * a->dtau));
+
+    a->delta         = (double*)malloc0 (a->msize * sizeof(double));
+    a->delta_copy    = (double*)malloc0 (a->msize * sizeof(double));
+    a->cfc_gain_copy = (double*)malloc0 (a->msize * sizeof(double));
+}
+
+void decalc_cfcomp(CFCOMP a)
+{
+    int i;
+    _aligned_free (a->cfc_gain_copy);
+    _aligned_free (a->delta_copy);
+    _aligned_free (a->delta);
+    _aligned_free (a->peq);
+    _aligned_free (a->comp);
+    _aligned_free (a->ep);
+    _aligned_free (a->gp);
+    _aligned_free (a->fp);
+
+    fftw_destroy_plan(a->Rrev);
+    fftw_destroy_plan(a->Rfor);
+    _aligned_free(a->outaccum);
+    for (i = 0; i < a->ovrlp; i++)
+        _aligned_free(a->save[i]);
+    _aligned_free(a->save);
+    _aligned_free(a->revfftout);
+    _aligned_free(a->revfftin);
+    _aligned_free(a->cfc_gain);
+    _aligned_free(a->mask);
+    _aligned_free(a->cmask);
+    _aligned_free(a->forfftout);
+    _aligned_free(a->forfftin);
+    _aligned_free(a->inaccum);
+    _aligned_free(a->window);
+}
+
+CFCOMP create_cfcomp (int run, int position, int peq_run, int size, double* in, double* out, int fsize, int ovrlp,
+    int rate, int wintype, int comp_method, int nfreqs, double precomp, double prepeq, double* F, double* G, double* E, double mtau, double dtau)
+{
+    CFCOMP a = (CFCOMP) malloc0 (sizeof (cfcomp));
+
+    a->run = run;
+    a->position = position;
+    a->peq_run = peq_run;
+    a->bsize = size;
+    a->in = in;
+    a->out = out;
+    a->fsize = fsize;
+    a->ovrlp = ovrlp;
+    a->rate = rate;
+    a->wintype = wintype;
+    a->comp_method = comp_method;
+    a->nfreqs = nfreqs;
+    a->precomp = precomp;
+    a->prepeq = prepeq;
+    a->mtau = mtau;                 // compression metering time constant
+    a->dtau = dtau;                 // compression display time constant
+    a->F = (double *)malloc0 (a->nfreqs * sizeof (double));
+    a->G = (double *)malloc0 (a->nfreqs * sizeof (double));
+    a->E = (double *)malloc0 (a->nfreqs * sizeof (double));
+    memcpy (a->F, F, a->nfreqs * sizeof (double));
+    memcpy (a->G, G, a->nfreqs * sizeof (double));
+    memcpy (a->E, E, a->nfreqs * sizeof (double));
+    calc_cfcomp (a);
+    return a;
+}
+
+void flush_cfcomp (CFCOMP a)
+{
+    int i;
+    memset (a->inaccum, 0, a->iasize * sizeof (double));
+    for (i = 0; i < a->ovrlp; i++)
+        memset (a->save[i], 0, a->fsize * sizeof (double));
+    memset (a->outaccum, 0, a->oasize * sizeof (double));
+    a->nsamps   = 0;
+    a->iainidx  = 0;
+    a->iaoutidx = 0;
+    a->oainidx  = a->init_oainidx;
+    a->oaoutidx = 0;
+    a->saveidx  = 0;
+    a->gain = 0.0;
+    memset(a->delta, 0, a->msize * sizeof(double));
+}
+
+void destroy_cfcomp (CFCOMP a)
+{
+    decalc_cfcomp (a);
+    _aligned_free (a->E);
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    _aligned_free (a);
+}
+
+
+void calc_mask (CFCOMP a)
+{
+    int i;
+    double comp, mask, delta;
+    switch (a->comp_method)
+    {
+    case 0:
+        {
+            double mag, test;
+            for (i = 0; i < a->msize; i++)
+            {
+                mag = sqrt (a->forfftout[2 * i + 0] * a->forfftout[2 * i + 0]
+                          + a->forfftout[2 * i + 1] * a->forfftout[2 * i + 1]);
+                comp = a->cfc_gain[i];
+                test = comp * mag;
+                if (test > 1.0)
+                    mask = 1.0 / mag;
+                else
+                    mask = comp;
+                a->cmask[i] = mask;
+                if (test > a->gain) a->gain = test;
+                else a->gain = a->mmult * a->gain;
+
+                delta = a->cfc_gain[i] - a->cmask[i];
+                if (delta > a->delta[i]) a->delta[i] = delta;
+                else a->delta[i] *= a->dmult;
+            }
+            break;
+        }
+    }
+    if (a->peq_run)
+    {
+        for (i = 0; i < a->msize; i++)
+        {
+            a->mask[i] = a->cmask[i] * a->prepeqlin * a->peq[i];
+        }
+    }
+    else
+        memcpy (a->mask, a->cmask, a->msize * sizeof (double));
+    // print_impulse ("mask.txt", a->msize, a->mask, 0, 0);
+    a->mask_ready = 1;
+}
+
+void xcfcomp (CFCOMP a, int pos)
+{
+    if (a->run && pos == a->position)
+    {
+        int i, j, k, sbuff, sbegin;
+        for (i = 0; i < 2 * a->bsize; i += 2)
+        {
+            a->inaccum[a->iainidx] = a->in[i];
+            a->iainidx = (a->iainidx + 1) % a->iasize;
+        }
+        a->nsamps += a->bsize;
+        while (a->nsamps >= a->fsize)
+        {
+            for (i = 0, j = a->iaoutidx; i < a->fsize; i++, j = (j + 1) % a->iasize)
+                a->forfftin[i] = a->pregain * a->window[i] * a->inaccum[j];
+            a->iaoutidx = (a->iaoutidx + a->incr) % a->iasize;
+            a->nsamps -= a->incr;
+            fftw_execute (a->Rfor);
+            calc_mask(a);
+            for (i = 0; i < a->msize; i++)
+            {
+                a->revfftin[2 * i + 0] = a->mask[i] * a->forfftout[2 * i + 0];
+                a->revfftin[2 * i + 1] = a->mask[i] * a->forfftout[2 * i + 1];
+            }
+            fftw_execute (a->Rrev);
+            for (i = 0; i < a->fsize; i++)
+                a->save[a->saveidx][i] = a->postgain * a->window[i] * a->revfftout[i];
+            for (i = a->ovrlp; i > 0; i--)
+            {
+                sbuff = (a->saveidx + i) % a->ovrlp;
+                sbegin = a->incr * (a->ovrlp - i);
+                for (j = sbegin, k = a->oainidx; j < a->incr + sbegin; j++, k = (k + 1) % a->oasize)
+                {
+                    if ( i == a->ovrlp)
+                        a->outaccum[k]  = a->save[sbuff][j];
+                    else
+                        a->outaccum[k] += a->save[sbuff][j];
+                }
+            }
+            a->saveidx = (a->saveidx + 1) % a->ovrlp;
+            a->oainidx = (a->oainidx + a->incr) % a->oasize;
+        }
+        for (i = 0; i < a->bsize; i++)
+        {
+            a->out[2 * i + 0] = a->outaccum[a->oaoutidx];
+            a->out[2 * i + 1] = 0.0;
+            a->oaoutidx = (a->oaoutidx + 1) % a->oasize;
+        }
+    }
+    else if (a->out != a->in)
+        memcpy (a->out, a->in, a->bsize * sizeof (complex));
+}
+
+void setBuffers_cfcomp (CFCOMP a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+}
+
+void setSamplerate_cfcomp (CFCOMP a, int rate)
+{
+    decalc_cfcomp (a);
+    a->rate = rate;
+    calc_cfcomp (a);
+}
+
+void setSize_cfcomp (CFCOMP a, int size)
+{
+    decalc_cfcomp (a);
+    a->bsize = size;
+    calc_cfcomp (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXACFCOMPRun (int channel, int run)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    if (a->run != run)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->run = run;
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetTXACFCOMPPosition (int channel, int pos)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    if (a->position != pos)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->position = pos;
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetTXACFCOMPprofile (int channel, int nfreqs, double* F, double* G, double *E)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a->nfreqs = nfreqs;
+    _aligned_free (a->E);
+    _aligned_free (a->F);
+    _aligned_free (a->G);
+    a->F = (double *)malloc0 (a->nfreqs * sizeof (double));
+    a->G = (double *)malloc0 (a->nfreqs * sizeof (double));
+    a->E = (double *)malloc0 (a->nfreqs * sizeof (double));
+    memcpy (a->F, F, a->nfreqs * sizeof (double));
+    memcpy (a->G, G, a->nfreqs * sizeof (double));
+    memcpy (a->E, E, a->nfreqs * sizeof (double));
+    _aligned_free (a->ep);
+    _aligned_free (a->gp);
+    _aligned_free (a->fp);
+    a->fp = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    a->gp = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    a->ep = (double *) malloc0 ((a->nfreqs + 2) * sizeof (double));
+    calc_comp(a);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXACFCOMPPrecomp (int channel, double precomp)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    if (a->precomp != precomp)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->precomp = precomp;
+        a->precomplin = pow (10.0, 0.05 * a->precomp);
+        for (int i = 0; i < a->msize; i++)
+        {
+            a->cfc_gain[i] = a->precomplin * a->comp[i];
+        }
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetTXACFCOMPPeqRun (int channel, int run)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    if (a->peq_run != run)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        a->peq_run = run;
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT
+void SetTXACFCOMPPrePeq (int channel, double prepeq)
+{
+    CFCOMP a = txa[channel].cfcomp.p;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a->prepeq = prepeq;
+    a->prepeqlin = pow (10.0, 0.05 * a->prepeq);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void GetTXACFCOMPDisplayCompression (int channel, double* comp_values, int* ready)
+{
+    int i;
+    CFCOMP a = txa[channel].cfcomp.p;
+    EnterCriticalSection(&ch[channel].csDSP);
+    if (*ready = a->mask_ready)
+    {
+        memcpy(a->delta_copy, a->delta, a->msize * sizeof(double));
+        memcpy(a->cfc_gain_copy, a->cfc_gain, a->msize * sizeof(double));
+        a->mask_ready = 0;
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+    if (*ready)
+    {
+        for (i = 0; i < a->msize; i++)
+            comp_values[i] = 20.0 * mlog10 (a->cfc_gain_copy[i] / (a->cfc_gain_copy[i] - a->delta_copy[i]));
+    }
+}

wdsp/cfcomp.h

@@ -0,0 +1,112 @@
+/*  cfcomp.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2017, 2021 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 _cfcomp_h
+#define _cfcomp_h
+
+typedef struct _cfcomp
+{
+    int run;
+    int position;
+    int bsize;
+    double* in;
+    double* out;
+    int fsize;
+    int ovrlp;
+    int incr;
+    double* window;
+    int iasize;
+    double* inaccum;
+    double* forfftin;
+    double* forfftout;
+    int msize;
+    double* cmask;
+    double* mask;
+    int mask_ready;
+    double* cfc_gain;
+    double* revfftin;
+    double* revfftout;
+    double** save;
+    int oasize;
+    double* outaccum;
+    double rate;
+    int wintype;
+    double pregain;
+    double postgain;
+    int nsamps;
+    int iainidx;
+    int iaoutidx;
+    int init_oainidx;
+    int oainidx;
+    int oaoutidx;
+    int saveidx;
+    fftw_plan Rfor;
+    fftw_plan Rrev;
+
+    int comp_method;
+    int nfreqs;
+    double* F;
+    double* G;
+    double* E;
+    double* fp;
+    double* gp;
+    double* ep;
+    double* comp;
+    double precomp;
+    double precomplin;
+    double* peq;
+    int peq_run;
+    double prepeq;
+    double prepeqlin;
+    double winfudge;
+
+    double gain;
+    double mtau;
+    double mmult;
+    // display stuff
+    double dtau;
+    double dmult;
+    double* delta;
+    double* delta_copy;
+    double* cfc_gain_copy;
+}cfcomp, *CFCOMP;
+
+extern CFCOMP create_cfcomp (int run, int position, int peq_run, int size, double* in, double* out, int fsize, int ovrlp,
+    int rate, int wintype, int comp_method, int nfreqs, double precomp, double prepeq, double* F, double* G, double* E, double mtau, double dtau);
+
+extern void destroy_cfcomp (CFCOMP a);
+
+extern void flush_cfcomp (CFCOMP a);
+
+extern void xcfcomp (CFCOMP a, int pos);
+
+extern void setBuffers_cfcomp (CFCOMP a, double* in, double* out);
+
+extern void setSamplerate_cfcomp (CFCOMP a, int rate);
+
+extern void setSize_cfcomp (CFCOMP a, int size);
+
+#endif

wdsp/cfir.c

@@ -0,0 +1,254 @@
+/*  cfir.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014, 2016, 2021 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
+
+*/
+#define _CRT_SECURE_NO_DEPRECATE
+#include "comm.h"
+
+void calc_cfir (CFIR a)
+{
+    double* impulse;
+    a->scale = 1.0 / (double)(2 * a->size);
+    impulse = cfir_impulse (a->nc, a->DD, a->R, a->Pairs, a->runrate, a->cicrate, a->cutoff, a->xtype, a->xbw, 1, a->scale, a->wintype);
+    a->p = create_fircore (a->size, a->in, a->out, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+}
+
+void decalc_cfir (CFIR a)
+{
+    destroy_fircore (a->p);
+}
+
+CFIR create_cfir (int run, int size, int nc, int mp, double* in, double* out, int runrate, int cicrate,
+    int DD, int R, int Pairs, double cutoff, int xtype, double xbw, int wintype)
+//  run:  0 - no action; 1 - operate
+//  size:  number of complex samples in an input buffer to the CFIR filter
+//  nc:  number of filter coefficients
+//  mp:  minimum phase flag
+//  in:  pointer to the input buffer
+//  out:  pointer to the output buffer
+//  rate:  samplerate
+//  DD:  differential delay of the CIC to be compensated (usually 1 or 2)
+//  R:  interpolation factor of CIC
+//  Pairs:  number of comb-integrator pairs in the CIC
+//  cutoff:  cutoff frequency
+//  xtype:  0 - fourth power transition; 1 - raised cosine transition; 2 - brick wall
+//  xbw:  width of raised cosine transition
+{
+    CFIR a = (CFIR) malloc0 (sizeof (cfir));
+    a->run = run;
+    a->size = size;
+    a->nc = nc;
+    a->mp = mp;
+    a->in = in;
+    a->out = out;
+    a->runrate = runrate;
+    a->cicrate = cicrate;
+    a->DD = DD;
+    a->R = R;
+    a->Pairs = Pairs;
+    a->cutoff = cutoff;
+    a->xtype = xtype;
+    a->xbw = xbw;
+    a->wintype = wintype;
+    calc_cfir (a);
+    return a;
+}
+
+void destroy_cfir (CFIR a)
+{
+    decalc_cfir (a);
+    _aligned_free (a);
+}
+
+void flush_cfir (CFIR a)
+{
+    flush_fircore (a->p);
+}
+
+void xcfir (CFIR a)
+{
+    if (a->run)
+        xfircore (a->p);
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_cfir (CFIR a, double* in, double* out)
+{
+    decalc_cfir (a);
+    a->in = in;
+    a->out = out;
+    calc_cfir (a);
+}
+
+void setSamplerate_cfir (CFIR a, int rate)
+{
+    decalc_cfir (a);
+    a->runrate = rate;
+    calc_cfir (a);
+}
+
+void setSize_cfir (CFIR a, int size)
+{
+    decalc_cfir (a);
+    a->size = size;
+    calc_cfir (a);
+}
+
+void setOutRate_cfir (CFIR a, int rate)
+{
+    decalc_cfir (a);
+    a->cicrate = rate;
+    calc_cfir (a);
+}
+
+double* cfir_impulse (int N, int DD, int R, int Pairs, double runrate, double cicrate, double cutoff, int xtype, double xbw, int rtype, double scale, int wintype)
+{
+    // N:       number of impulse response samples
+    // DD:      differential delay used in the CIC filter
+    // R:       interpolation / decimation factor of the CIC
+    // Pairs:   number of comb-integrator pairs in the CIC
+    // runrate: sample rate at which this filter is to run (assumes there may be flat interp. between this filter and the CIC)
+    // cicrate: sample rate at interface to CIC
+    // cutoff:  cutoff frequency
+    // xtype:   transition type, 0 for 4th-power rolloff, 1 for raised cosine, 2 for brick wall
+    // xbw:     transition bandwidth for raised cosine
+    // rtype:   0 for real output, 1 for complex output
+    // scale:   scale factor to be applied to the output
+    int i, j;
+    double tmp, local_scale, ri, mag, fn;
+    double* impulse;
+    double* A = (double *) malloc0 (N * sizeof (double));
+    double ft = cutoff / cicrate;                                       // normalized cutoff frequency
+    int u_samps = (N + 1) / 2;                                          // number of unique samples,  OK for odd or even N
+    int c_samps = (int)(cutoff / runrate * N) + (N + 1) / 2 - N / 2;    // number of unique samples within bandpass, OK for odd or even N
+    int x_samps = (int)(xbw / runrate * N);                             // number of unique samples in transition region, OK for odd or even N
+    double offset = 0.5 - 0.5 * (double)((N + 1) / 2 - N / 2);          // sample offset from center, OK for odd or even N
+    double* xistion = (double *) malloc0 ((x_samps + 1) * sizeof (double));
+    double delta = PI / (double)x_samps;
+    double L = cicrate / runrate;
+    double phs = 0.0;
+    for (i = 0; i <= x_samps; i++)
+    {
+        xistion[i] = 0.5 * (cos (phs) + 1.0);
+        phs += delta;
+    }
+    if ((tmp = DD * R * sin (PI * ft / R) / sin (PI * DD * ft)) < 0.0)  //normalize by peak gain
+        tmp = -tmp;
+    local_scale = scale / pow (tmp, Pairs);
+    if (xtype == 0)
+    {
+        for (i = 0, ri = offset; i < u_samps; i++, ri += 1.0)
+        {
+            fn = ri / (L * (double)N);
+            if (fn <= ft)
+            {
+                if (fn == 0.0) tmp = 1.0;
+                else if ((tmp = DD * R * sin (PI * fn / R) / sin (PI * DD * fn)) < 0.0)
+                    tmp = -tmp;
+                mag = pow (tmp, Pairs) * local_scale;
+            }
+            else
+                mag *= (ft * ft * ft * ft) / (fn * fn * fn * fn);
+            A[i] = mag;
+        }
+    }
+    else if (xtype == 1)
+    {
+        for (i = 0, ri = offset; i < u_samps; i++, ri += 1.0)
+        {
+            fn = ri / (L *(double)N);
+            if (i < c_samps)
+            {
+                if (fn == 0.0) tmp = 1.0;
+                else if ((tmp = DD * R * sin (PI * fn / R) / sin (PI * DD * fn)) < 0.0)
+                    tmp = -tmp;
+                mag = pow (tmp, Pairs) * local_scale;
+                A[i] = mag;
+            }
+            else if ( i >= c_samps && i <= c_samps + x_samps)
+                A[i] = mag * xistion[i - c_samps];
+            else
+                A[i] = 0.0;
+        }
+    }
+    else if (xtype == 2)
+    {
+        for (i = 0, ri = offset; i < u_samps; i++, ri += 1.0)
+        {
+            fn = ri / (L * (double)N);
+            if (fn <= ft)
+            {
+                if (fn == 0.0) tmp = 1.0;
+                else if ((tmp = DD * R * sin(PI * fn / R) / sin(PI * DD * fn)) < 0.0)
+                    tmp = -tmp;
+                mag = pow (tmp, Pairs) * local_scale;
+            }
+            else
+                mag = 0.0;
+            A[i] = mag;
+        }
+    }
+    if (N & 1)
+        for (i = u_samps, j = 2; i < N; i++, j++)
+            A[i] = A[u_samps - j];
+    else
+        for (i = u_samps, j = 1; i < N; i++, j++)
+            A[i] = A[u_samps - j];
+    impulse = fir_fsamp (N, A, rtype, 1.0, wintype);
+    // print_impulse ("cfirImpulse.txt", N, impulse, 1, 0);
+    _aligned_free (A);
+    return impulse;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXACFIRRun (int channel, int run)
+{
+    EnterCriticalSection(&ch[channel].csDSP);
+    txa[channel].cfir.p->run = run;
+    LeaveCriticalSection(&ch[channel].csDSP);
+}
+
+PORT
+void SetTXACFIRNC(int channel, int nc)
+{
+    // NOTE:  'nc' must be >= 'size'
+    CFIR a;
+    EnterCriticalSection(&ch[channel].csDSP);
+    a = txa[channel].cfir.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        decalc_cfir(a);
+        calc_cfir(a);
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+}

wdsp/cfir.h

@@ -0,0 +1,75 @@
+/*  cfir.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014, 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 _cfir_h
+#define _cfir_h
+#include "firmin.h"
+typedef struct _cfir
+{
+    int run;
+    int size;
+    int nc;
+    int mp;
+    double* in;
+    double* out;
+    int runrate;
+    int cicrate;
+    int DD;
+    int R;
+    int Pairs;
+    double cutoff;
+    double scale;
+    int xtype;
+    double xbw;
+    int wintype;
+    FIRCORE p;
+} cfir, *CFIR;
+
+extern CFIR create_cfir (int run, int size, int nc, int mp, double* in, double* out, int runrate, int cicrate,
+    int DD, int R, int Pairs, double cutoff, int xtype, double xbw, int wintype);
+
+extern void destroy_cfir (CFIR a);
+
+extern void flush_cfir (CFIR a);
+
+extern void xcfir (CFIR a);
+
+extern void setBuffers_cfir (CFIR a, double* in, double* out);
+
+extern void setSamplerate_cfir (CFIR a, int rate);
+
+extern void setSize_cfir (CFIR a, int size);
+
+extern void setOutRate_cfir (CFIR a, int rate);
+
+extern double* cfir_impulse (int N, int DD, int R, int Pairs, double runrate, double cicrate,
+    double cutoff, int xtype, double xbw, int rtype, double scale, int wintype);
+
+extern __declspec (dllexport) void SetTXACFIRRun(int channel, int run);
+
+extern __declspec (dllexport) void SetTXACFIRNC(int channel, int nc);
+
+#endif

wdsp/channel.c

@@ -0,0 +1,347 @@
+/*  channel.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"
+
+struct _ch ch[MAX_CHANNELS];
+
+void start_thread (int channel)
+{
+    HANDLE handle = (HANDLE) _beginthread(wdspmain, 0, (void *)(uintptr_t)channel);
+    //SetThreadPriority(handle, THREAD_PRIORITY_HIGHEST);
+}
+
+void pre_main_build (int channel)
+{
+    if (ch[channel].in_rate  >= ch[channel].dsp_rate)
+        ch[channel].dsp_insize  = ch[channel].dsp_size * (ch[channel].in_rate  / ch[channel].dsp_rate);
+    else
+        ch[channel].dsp_insize  = ch[channel].dsp_size / (ch[channel].dsp_rate /  ch[channel].in_rate);
+
+    if (ch[channel].out_rate >= ch[channel].dsp_rate)
+        ch[channel].dsp_outsize = ch[channel].dsp_size * (ch[channel].out_rate / ch[channel].dsp_rate);
+    else
+        ch[channel].dsp_outsize = ch[channel].dsp_size / (ch[channel].dsp_rate / ch[channel].out_rate);
+
+    if (ch[channel].in_rate  >= ch[channel].out_rate)
+        ch[channel].out_size    = ch[channel].in_size  / (ch[channel].in_rate  / ch[channel].out_rate);
+    else
+        ch[channel].out_size    = ch[channel].in_size  * (ch[channel].out_rate /  ch[channel].in_rate);
+
+    InitializeCriticalSectionAndSpinCount ( &ch[channel].csDSP, 2500 );
+    InitializeCriticalSectionAndSpinCount ( &ch[channel].csEXCH,  2500 );
+    InterlockedBitTestAndReset (&ch[channel].flushflag, 0);
+    create_iobuffs (channel);
+}
+
+void post_main_build (int channel)
+{
+    InterlockedBitTestAndSet (&ch[channel].run, 0);
+    start_thread (channel);
+    if (ch[channel].state == 1)
+        InterlockedBitTestAndSet (&ch[channel].exchange, 0);
+}
+
+void build_channel (int channel)
+{
+    pre_main_build (channel);
+    create_main (channel);
+    post_main_build (channel);
+}
+
+PORT
+void OpenChannel (int channel, int in_size, int dsp_size, int input_samplerate, int dsp_rate, int output_samplerate,
+    int type, int state, double tdelayup, double tslewup, double tdelaydown, double tslewdown, int bfo)
+{
+    ch[channel].in_size = in_size;
+    ch[channel].dsp_size = dsp_size;
+    ch[channel].in_rate = input_samplerate;
+    ch[channel].dsp_rate = dsp_rate;
+    ch[channel].out_rate = output_samplerate;
+    ch[channel].type = type;
+    ch[channel].state = state;
+    ch[channel].tdelayup = tdelayup;
+    ch[channel].tslewup = tslewup;
+    ch[channel].tdelaydown = tdelaydown;
+    ch[channel].tslewdown = tslewdown;
+    ch[channel].bfo = bfo;
+    InterlockedBitTestAndReset (&ch[channel].exchange, 0);
+    build_channel (channel);
+    if (ch[channel].state)
+    {
+        InterlockedBitTestAndSet (&ch[channel].iob.pc->slew.upflag, 0);
+        InterlockedBitTestAndSet (&ch[channel].iob.ch_upslew, 0);
+        InterlockedBitTestAndReset (&ch[channel].iob.pc->exec_bypass, 0);
+        InterlockedBitTestAndSet (&ch[channel].exchange, 0);
+    }
+#if !defined(linux) && !defined(__APPLE__)
+    _MM_SET_FLUSH_ZERO_MODE (_MM_FLUSH_ZERO_ON);
+#endif
+}
+
+void pre_main_destroy (int channel)
+{
+    IOB a = ch[channel].iob.pc;
+    InterlockedBitTestAndReset (&ch[channel].exchange, 0);
+    InterlockedBitTestAndReset (&ch[channel].run, 0);
+    InterlockedBitTestAndSet (&ch[channel].iob.pc->exec_bypass, 0);
+    ReleaseSemaphore (a->Sem_BuffReady, 1, 0);
+    Sleep (25);
+}
+
+void post_main_destroy (int channel)
+{
+    destroy_iobuffs (channel);
+    DeleteCriticalSection ( &ch[channel].csEXCH  );
+    DeleteCriticalSection ( &ch[channel].csDSP );
+}
+
+PORT
+void CloseChannel (int channel)
+{
+    pre_main_destroy (channel);
+    destroy_main (channel);
+    post_main_destroy (channel);
+}
+
+void flushChannel (void* p)
+{
+    int channel = (int)(uintptr_t)p;
+        IOB a = ch[channel].iob.pc;
+        while (!InterlockedAnd(&a->flush_bypass, 0xffffffff))
+        {
+                WaitForSingleObject(a->Sem_Flush, INFINITE);
+                if (!InterlockedAnd(&a->flush_bypass, 0xffffffff))
+                {
+                        EnterCriticalSection(&ch[channel].csDSP);
+                        EnterCriticalSection(&ch[channel].csEXCH);
+                        flush_iobuffs(channel);
+                        InterlockedBitTestAndSet(&a->exec_bypass, 0);
+                        flush_main(channel);
+                        LeaveCriticalSection(&ch[channel].csEXCH);
+                        LeaveCriticalSection(&ch[channel].csDSP);
+                        InterlockedBitTestAndReset(&ch[channel].flushflag, 0);
+                }
+        }
+        InterlockedBitTestAndReset(&a->flush_bypass, 0);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       Channel Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetType (int channel, int type)
+{   // no need to rebuild buffers; but we did anyway
+    if (type != ch[channel].type)
+    {
+        CloseChannel (channel);
+        ch[channel].type = type;
+        build_channel (channel);
+    }
+}
+
+PORT
+void SetInputBuffsize (int channel, int in_size)
+{   // we do not rebuild main here since it didn't change
+    if (in_size != ch[channel].in_size)
+    {
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].in_size = in_size;
+        pre_main_build (channel);
+        post_main_build (channel);
+    }
+}
+
+PORT
+void SetDSPBuffsize (int channel, int dsp_size)
+{
+    if (dsp_size != ch[channel].dsp_size)
+    {
+        int oldstate = SetChannelState (channel, 0, 1);
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].dsp_size = dsp_size;
+        pre_main_build (channel);
+        setDSPBuffsize_main (channel);
+        post_main_build (channel);
+        SetChannelState (channel, oldstate, 0);
+    }
+}
+
+PORT
+void SetInputSamplerate (int channel, int in_rate)
+{   // no re-build of main required
+    if (in_rate != ch[channel].in_rate)
+    {
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].in_rate = in_rate;
+        pre_main_build (channel);
+        setInputSamplerate_main (channel);
+        post_main_build (channel);
+    }
+}
+
+PORT
+void SetDSPSamplerate (int channel, int dsp_rate)
+{
+    if (dsp_rate != ch[channel].dsp_rate)
+    {
+        int oldstate = SetChannelState (channel, 0, 1);
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].dsp_rate = dsp_rate;
+        pre_main_build (channel);
+        setDSPSamplerate_main (channel);
+        post_main_build (channel);
+        SetChannelState (channel, oldstate, 0);
+    }
+}
+
+PORT
+void SetOutputSamplerate (int channel, int out_rate)
+{   // no re-build of main required
+    if (out_rate != ch[channel].out_rate)
+    {
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].out_rate = out_rate;
+        pre_main_build (channel);
+        setOutputSamplerate_main (channel);
+        post_main_build (channel);
+    }
+}
+
+PORT
+void SetAllRates (int channel, int in_rate, int dsp_rate, int out_rate)
+{
+    if ((in_rate != ch[channel].in_rate) || (dsp_rate != ch[channel].dsp_rate) || (out_rate != ch[channel].out_rate))
+    {
+        pre_main_destroy (channel);
+        post_main_destroy (channel);
+        ch[channel].in_rate  = in_rate;
+        ch[channel].dsp_rate = dsp_rate;
+        ch[channel].out_rate = out_rate;
+        pre_main_build (channel);
+        setInputSamplerate_main (channel);
+        setDSPSamplerate_main (channel);
+        setOutputSamplerate_main (channel);
+        post_main_build (channel);
+    }
+}
+
+PORT
+int SetChannelState (int channel, int state, int dmode)
+{
+    IOB a = ch[channel].iob.pc;
+    int prior_state = ch[channel].state;
+    int count = 0;
+    const int timeout = 100;
+    if (ch[channel].state != state)
+    {
+        ch[channel].state = state;
+        switch (ch[channel].state)
+        {
+        case 0:
+            InterlockedBitTestAndSet (&a->slew.downflag, 0);
+            InterlockedBitTestAndSet (&ch[channel].flushflag, 0);
+            if (dmode)
+            {
+                while (_InterlockedAnd (&ch[channel].flushflag, 1) && count < timeout)
+                {
+                    Sleep(1);
+                    count++;
+                }
+            }
+            if (count >= timeout)
+            {
+                InterlockedBitTestAndReset (&ch[channel].exchange, 0);
+                InterlockedBitTestAndReset (&ch[channel].flushflag, 0);
+                InterlockedBitTestAndReset (&a->slew.downflag, 0);
+            }
+            break;
+        case 1:
+            InterlockedBitTestAndSet (&a->slew.upflag, 0);
+            InterlockedBitTestAndSet (&ch[channel].iob.ch_upslew, 0);
+            InterlockedBitTestAndReset (&ch[channel].iob.pc->exec_bypass, 0);
+            InterlockedBitTestAndSet (&ch[channel].exchange, 0);
+            break;
+        }
+    }
+    return prior_state;
+}
+
+PORT
+void SetChannelTDelayUp (int channel, double time)
+{
+    IOB a;
+    EnterCriticalSection (&ch[channel].csEXCH);
+    a = ch[channel].iob.pc;
+    ch[channel].tdelayup = time;
+    a->slew.ndelup = (int)(ch[a->channel].tdelayup * ch[a->channel].in_rate);
+    flush_slews (a);
+    LeaveCriticalSection (&ch[channel].csEXCH);
+}
+
+PORT
+void SetChannelTSlewUp (int channel, double time)
+{
+    IOB a;
+    EnterCriticalSection (&ch[channel].csEXCH);
+    a = ch[channel].iob.pc;
+    ch[channel].tslewup = time;
+    destroy_slews (a);
+    create_slews (a);
+    LeaveCriticalSection (&ch[channel].csEXCH);
+}
+
+PORT
+void SetChannelTDelayDown (int channel, double time)
+{
+    IOB a;
+    EnterCriticalSection (&ch[channel].csEXCH);
+    a = ch[channel].iob.pc;
+    ch[channel].tdelaydown = time;
+    a->slew.ndeldown = (int)(ch[a->channel].tdelaydown * ch[a->channel].out_rate);
+    flush_slews (a);
+    LeaveCriticalSection (&ch[channel].csEXCH);
+}
+
+PORT
+void SetChannelTSlewDown (int channel, double time)
+{
+    IOB a;
+    EnterCriticalSection (&ch[channel].csEXCH);
+    a = ch[channel].iob.pc;
+    ch[channel].tslewdown = time;
+    destroy_slews (a);
+    create_slews (a);
+    LeaveCriticalSection (&ch[channel].csEXCH);
+}

wdsp/channel.h

@@ -0,0 +1,84 @@
+/*  channel.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 _setupchannel_h
+#define _setupchannel_h
+#include "comm.h"
+
+struct _ch
+{
+    int type;
+    volatile long run;          // when 1, thread loops; when 0, thread terminates
+    volatile long exchange;     // when 1, fexchange() operates; when 0, it just returns
+    int in_rate;                // input samplerate
+    int out_rate;               // output samplerate
+    int in_size;                // input buffsize (complex samples) in a fexchange() operation
+    int dsp_rate;               // sample rate for mainstream dsp processing
+    int dsp_size;               // number complex samples processed per buffer in mainstream dsp processing
+    int dsp_insize;             // size (complex samples) of the output of the r1 (input) buffer
+    int dsp_outsize;            // size (complex samples) of the input of the r2 (output) buffer
+    int out_size;               // output buffsize (complex samples) in a fexchange() operation
+    CRITICAL_SECTION csDSP;     // used to block dsp while parameters are updated or buffers flushed
+    CRITICAL_SECTION csEXCH;    // used to block fexchange() while parameters are updated or buffers flushed
+    int state;                  // 0 for channel OFF; 1 for channel ON
+    double tdelayup;
+    double tslewup;
+    double tdelaydown;
+    double tslewdown;
+    int bfo;                    // 'block_for_output', block fexchange until output is available
+    volatile long flushflag;
+    struct  //io buffers
+    {
+        IOB pc, pd, pe, pf;     // copies for console calls, dsp, exchange, and flush thread
+        volatile long ch_upslew;
+    } iob;
+};
+
+extern struct _ch ch[];
+
+PORT void OpenChannel (int channel, int in_size, int dsp_size, int input_samplerate, int dsp_rate, int output_samplerate, int type, int state, double tdelayup, double tslewup, double tdelaydown, double tslewdown, int bfo);
+
+PORT void CloseChannel (int channel);
+
+extern void flushChannel (void* p);
+
+PORT void SetType (int channel, int type);
+
+PORT void SetInputBuffsize (int channel, int in_size);
+
+PORT void SetDSPBuffsize (int channel, int dsp_size);
+
+PORT void SetInputSamplerate  (int channel, int samplerate);
+
+PORT void SetDSPSamplerate (int channel, int samplerate);
+
+PORT void SetOutputSamplerate (int channel, int samplerate);
+
+PORT void SetAllRates (int channel, int in_rate, int dsp_rate, int out_rate);
+
+PORT int SetChannelState (int channel, int state, int dmode);
+
+#endif

wdsp/comm.h

@@ -0,0 +1,151 @@
+/*  comm.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
+
+*/
+
+#if defined(linux) || defined(__APPLE__)
+#include <stdlib.h>
+#include <pthread.h>
+#include <semaphore.h>
+#include <string.h>
+#include "linux_port.h"
+#ifdef ANDROID
+#include <android/log.h>
+#define APPNAME "WDSP"
+#define LOGD(LOG_TAG, ...) __android_log_print(ANDROID_LOG_DEBUG, LOG_TAG, __VA_ARGS__)
+#define LOGI(LOG_TAG, ...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG,__VA_ARGS__)
+#define LOGV(LOG_TAG, ...) __android_log_print(ANDROID_LOG_VERBOSE, LOG_TAG, __VA_ARGS__)
+#define LOGW(LOG_TAG, ...) __android_log_print(ANDROID_LOG_WARN, LOG_TAG, __VA_ARGS__)
+#define LOGE(LOG_TAG, ...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#endif
+#endif
+
+#ifdef _WIN32
+#include <Windows.h>
+#include <process.h>
+#include <intrin.h>
+#endif
+#include <math.h>
+#include <stdint.h>
+#include <time.h>
+#ifdef _WIN32
+#include <avrt.h>
+#endif
+#include "fftw3.h"
+
+#include "amd.h"
+#include "ammod.h"
+#include "amsq.h"
+#include "analyzer.h"
+#include "anf.h"
+#include "anr.h"
+#include "bandpass.h"
+#include "calcc.h"
+#include "cblock.h"
+#include "cfcomp.h"
+#include "cfir.h"
+#include "channel.h"
+#include "compress.h"
+#include "delay.h"
+#include "dexp.h"
+#include "div.h"
+#include "eer.h"
+#include "emnr.h"
+#include "emph.h"
+#include "eq.h"
+#include "fcurve.h"
+#include "fir.h"
+#include "firmin.h"
+#include "fmd.h"
+#include "fmmod.h"
+#include "fmsq.h"
+#include "gain.h"
+#include "gen.h"
+#include "icfir.h"
+#include "iir.h"
+#include "iobuffs.h"
+#include "iqc.h"
+#include "lmath.h"
+#include "main.h"
+#include "meter.h"
+#include "meterlog10.h"
+#include "nbp.h"
+#include "nob.h"
+#include "nobII.h"
+#include "osctrl.h"
+#include "patchpanel.h"
+#include "resample.h"
+#include "rmatch.h"
+#include "RXA.h"
+#include "sender.h"
+#include "shift.h"
+#include "siphon.h"
+#include "slew.h"
+#include "snb.h"
+#include "ssql.h"
+#include "syncbuffs.h"
+#include "TXA.h"
+#include "utilities.h"
+#include "varsamp.h"
+#include "wcpAGC.h"
+
+// manage differences among consoles
+#define _Thetis
+
+// channel definitions
+#define MAX_CHANNELS                    32                  // maximum number of supported channels
+#define DSP_MULT                        2                   // number of dsp_buffsizes that are held in an iobuff pseudo-ring
+#define INREAL                          float               // data type for channel input buffer
+#define OUTREAL                         float               // data type for channel output buffer
+
+// display definitions
+#define dMAX_DISPLAYS                   64                  // maximum number of displays = max instances
+#define dMAX_STITCH                     4                   // maximum number of sub-spans to stitch together
+#define dMAX_NUM_FFT                    1                   // maximum number of ffts for an elimination
+#define dMAX_PIXELS                     16384               // maximum number of pixels that can be requested
+#define dMAX_AVERAGE                    60                  // maximum number of pixel frames that will be window-averaged
+#ifdef _Thetis
+#define dINREAL                         double
+#else
+#define dINREAL                         float
+#endif
+#define dOUTREAL                        float
+#define dSAMP_BUFF_MULT                 2                   // ratio of input sample buffer size to fft size (for overlap)
+#define dNUM_PIXEL_BUFFS                3                   // number of pixel output buffers
+#define dMAX_M                          1                   // number of variables to calibrate
+#define dMAX_N                          100                 // maximum number of frequencies at which to calibrate
+#define dMAX_CAL_SETS                   2                   // maximum number of calibration data sets
+#define dMAX_PIXOUTS                    4                   // maximum number of det/avg/outputs per display instance
+
+// wisdom definitions
+#define MAX_WISDOM_SIZE_DISPLAY         262144
+#define MAX_WISDOM_SIZE_FILTER          262144              // was 32769
+
+// math definitions
+#define PI                              3.1415926535897932
+#define TWOPI                           6.2831853071795864
+
+// miscellaneous
+typedef double complex[2];
+#define PORT                            __declspec( dllexport )

wdsp/compress.c

@@ -0,0 +1,117 @@
+/*  compress.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2011, 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
+
+This software is based upon the algorithm described by Peter Martinez, G3PLX,
+in the January 2010 issue of RadCom magazine.
+
+*/
+
+#include "comm.h"
+
+COMPRESSOR create_compressor (
+                int run,
+                int buffsize,
+                double* inbuff,
+                double* outbuff,
+                double gain )
+{
+    COMPRESSOR a;
+    a = (COMPRESSOR) malloc0 (sizeof (compressor));
+    a->run = run;
+    a->inbuff = inbuff;
+    a->outbuff = outbuff;
+    a->buffsize = buffsize;
+    a->gain = gain;
+    return a;
+}
+
+void destroy_compressor (COMPRESSOR a)
+{
+    _aligned_free (a);
+}
+
+void flush_compressor (COMPRESSOR a)
+{
+
+}
+
+void xcompressor (COMPRESSOR a)
+{
+    int i;
+    double mag;
+    if (a->run)
+        for (i = 0; i < a->buffsize; i++)
+        {
+            mag = sqrt(a->inbuff[2 * i + 0] * a->inbuff[2 * i + 0] + a->inbuff[2 * i + 1] * a->inbuff[2 * i + 1]);
+            if (a->gain * mag > 1.0)
+                a->outbuff[2 * i + 0] = a->inbuff[2 * i + 0] / mag;
+            else
+                a->outbuff[2 * i + 0] = a->inbuff[2 * i + 0] * a->gain;
+            a->outbuff[2 * i + 1] = 0.0;
+        }
+    else if (a->inbuff != a->outbuff)
+        memcpy(a->outbuff, a->inbuff, a->buffsize * sizeof (complex));
+}
+
+void setBuffers_compressor (COMPRESSOR a, double* in, double* out)
+{
+    a->inbuff = in;
+    a->outbuff = out;
+}
+
+void setSamplerate_compressor (COMPRESSOR a, int rate)
+{
+
+}
+
+void setSize_compressor (COMPRESSOR a, int size)
+{
+    a->buffsize = size;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT void
+SetTXACompressorRun (int channel, int run)
+{
+    if (txa[channel].compressor.p->run != run)
+    {
+        EnterCriticalSection (&ch[channel].csDSP);
+        txa[channel].compressor.p->run = run;
+        TXASetupBPFilters (channel);
+        LeaveCriticalSection (&ch[channel].csDSP);
+    }
+}
+
+PORT void
+SetTXACompressorGain (int channel, double gain)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].compressor.p->gain = pow (10.0, gain / 20.0);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/compress.h

@@ -0,0 +1,64 @@
+/*  compress.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2011, 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 _compressor_h
+#define _compressor_h
+
+typedef struct _compressor
+{
+    int run;
+    int buffsize;
+    double *inbuff;
+    double *outbuff;
+    double gain;
+} compressor, *COMPRESSOR;
+
+extern void xcompressor (COMPRESSOR a);
+
+extern COMPRESSOR create_compressor (
+                int run,
+                int buffsize,
+                double* inbuff,
+                double* outbuff,
+                double gain );
+
+extern void destroy_compressor (COMPRESSOR a);
+
+extern void flush_compressor (COMPRESSOR a);
+
+extern void setBuffers_compressor (COMPRESSOR a, double* in, double* out);
+
+extern void setSamplerate_compressor (COMPRESSOR a, int rate);
+
+extern void setSize_compressor (COMPRESSOR a, int size);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXACompressorRun (int channel, int run);
+
+extern __declspec (dllexport) void SetTXACompressorGain (int channel, double gain);
+
+#endif

wdsp/delay.c

@@ -0,0 +1,135 @@
+/*  delay.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2019 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"
+
+DELAY create_delay (int run, int size, double* in, double* out, int rate, double tdelta, double tdelay)
+{
+    DELAY a = (DELAY) malloc0 (sizeof (delay));
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = rate;
+    a->tdelta = tdelta;
+    a->tdelay = tdelay;
+    a->L = (int)(0.5 + 1.0 / (a->tdelta * (double)a->rate));
+    a->adelta = 1.0 / (a->rate * a->L);
+    a->ft = 0.45 / (double)a->L;
+    a->ncoef = (int)(60.0 / a->ft);
+    a->ncoef = (a->ncoef / a->L + 1) * a->L;
+    a->cpp = a->ncoef / a->L;
+    a->phnum = (int)(0.5 + a->tdelay / a->adelta);
+    a->snum = a->phnum / a->L;
+    a->phnum %= a->L;
+    a->idx_in = 0;
+    a->adelay = a->adelta * (a->snum * a->L + a->phnum);
+    a->h = fir_bandpass (a->ncoef,-a->ft, +a->ft, 1.0, 1, 0, (double)a->L);
+    a->rsize = a->cpp + (WSDEL - 1);
+    a->ring = (double *) malloc0 (a->rsize * sizeof (complex));
+    InitializeCriticalSectionAndSpinCount ( &a->cs_update, 2500 );
+    return a;
+}
+
+void destroy_delay (DELAY a)
+{
+    DeleteCriticalSection (&a->cs_update);
+    _aligned_free (a->ring);
+    _aligned_free (a->h);
+    _aligned_free (a);
+}
+
+void flush_delay (DELAY a)
+{
+    memset (a->ring, 0, a->cpp * sizeof (complex));
+    a->idx_in = 0;
+}
+
+void xdelay (DELAY a)
+{
+    EnterCriticalSection (&a->cs_update);
+    if (a->run)
+    {
+        int i, j, k, idx, n;
+        double Itmp, Qtmp;
+        for (i = 0; i < a->size; i++)
+        {
+            a->ring[2 * a->idx_in + 0] = a->in[2 * i + 0];
+            a->ring[2 * a->idx_in + 1] = a->in[2 * i + 1];
+            Itmp = 0.0;
+            Qtmp = 0.0;
+            if ((n = a->idx_in + a->snum) >= a->rsize) n -= a->rsize;
+            for (j = 0, k = a->L - 1 - a->phnum; j < a->cpp; j++, k+= a->L)
+            {
+                if ((idx = n + j) >= a->rsize) idx -= a->rsize;
+                Itmp += a->ring[2 * idx + 0] * a->h[k];
+                Qtmp += a->ring[2 * idx + 1] * a->h[k];
+            }
+            a->out[2 * i + 0] = Itmp;
+            a->out[2 * i + 1] = Qtmp;
+            if (--a->idx_in < 0) a->idx_in = a->rsize - 1;
+        }
+    }
+    else if (a->out != a->in)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+    LeaveCriticalSection (&a->cs_update);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                             Properties                                                *
+*                                                                                                       *
+********************************************************************************************************/
+
+void SetDelayRun (DELAY a, int run)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+double SetDelayValue (DELAY a, double tdelay)
+{
+    double adelay;
+    EnterCriticalSection (&a->cs_update);
+    a->tdelay = tdelay;
+    a->phnum = (int)(0.5 + a->tdelay / a->adelta);
+    a->snum = a->phnum / a->L;
+    a->phnum %= a->L;
+    a->adelay = a->adelta * (a->snum * a->L + a->phnum);
+    adelay = a->adelay;
+    LeaveCriticalSection (&a->cs_update);
+    return adelay;
+}
+
+void SetDelayBuffs (DELAY a, int size, double* in, double* out)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    LeaveCriticalSection (&a->cs_update);
+}

wdsp/delay.h

@@ -0,0 +1,77 @@
+/*  delay.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2019 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 _delay_h
+#define _delay_h
+
+#define WSDEL   1025    // number of supported whole sample delays
+
+typedef struct _delay
+{
+    int run;            // run
+    int size;           // number of input samples per buffer
+    double* in;         // input buffer
+    double* out;        // output buffer
+    int rate;           // samplerate
+    double tdelta;      // delay increment required (seconds)
+    double tdelay;      // delay requested (seconds)
+
+    int L;              // interpolation factor
+    int ncoef;          // number of coefficients
+    int cpp;            // coefficients per phase
+    double ft;          // normalized cutoff frequency
+    double* h;          // coefficients
+    int snum;           // starting sample number (0 for sub-sample delay)
+    int phnum;          // phase number
+
+    int idx_in;         // index for input into ring
+    int rsize;          // ring size in complex samples
+    double* ring;       // ring buffer
+
+    double adelta;      // actual delay increment
+    double adelay;      // actual delay
+
+    CRITICAL_SECTION cs_update;
+
+} delay, *DELAY;
+
+extern DELAY create_delay (int run, int size, double* in, double* out, int rate, double tdelta, double tdelay);
+
+extern void destroy_delay (DELAY a);
+
+extern void flush_delay (DELAY a);
+
+extern void xdelay (DELAY a);
+
+// Properties
+
+extern void SetDelayRun (DELAY a, int run);
+
+extern double SetDelayValue (DELAY a, double delay);        // returns actual delay in seconds
+
+extern void SetDelayBuffs (DELAY a, int size, double* in, double* out);
+
+#endif

wdsp/dexp.c

@@ -0,0 +1,716 @@
+/*  dexp.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2018, 2019 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"
+
+DEXP pdexp[4];
+
+DELRING calc_delring (int rsize, int size, int delay, double* in, double* out)
+{
+    DELRING a = (DELRING) malloc0 (sizeof (delring));
+    a->rsize = rsize;
+    a->size = size;
+    a->rdelay = delay;
+    a->in = in;
+    a->out = out;
+    a->ring = (double *) malloc0 (a->rsize * sizeof (complex));
+    a->inptr = a->rdelay;
+    a->outptr = 0;
+    return a;
+}
+
+void decalc_delring (DELRING a)
+{
+    _aligned_free (a->ring);
+    _aligned_free (a);
+}
+
+void flush_delring (DELRING a)
+{
+    memset (a->ring, 0, a->rsize * sizeof (complex));
+    a->inptr = a->rdelay;
+    a->outptr = 0;
+}
+
+void xdelring (DELRING a)
+{
+    int first, second;
+    // copy in
+    if (a->size > (a->rsize - a->inptr))
+    {
+        first = a->rsize - a->inptr;
+        second = a->size - first;
+    }
+    else
+    {
+        first = a->size;
+        second = 0;
+    }
+    memcpy (a->ring + 2 * a->inptr, a->in, first * sizeof (complex));
+    memcpy (a->ring, a->in + 2 * first, second * sizeof (complex));
+    a->inptr = (a->inptr + a->size) % a->rsize;
+    // copy out
+    if (a->size > (a->rsize - a->outptr))
+    {
+        first = a->rsize - a->outptr;
+        second = a->size - first;
+    }
+    else
+    {
+        first = a->size;
+        second = 0;
+    }
+    memcpy (a->out, a->ring + 2 * a->outptr, first * sizeof (complex));
+    memcpy (a->out + 2 * first, a->ring, second * sizeof (complex));
+    a->outptr = (a->outptr + a->size) % a->rsize;
+}
+
+void calc_slews (DEXP a)
+{
+    int i;
+    double delta, theta;
+    delta = PI / (double)a->nattack;
+    theta = 0.0;
+    for (i = 0; i <= a->nattack; i++)
+    {
+        a->cattack[i] = a->low_gain + (1.0 - a->low_gain) * 0.5 * (1.0 - cos (theta));
+        theta += delta;
+    }
+    delta = PI / (double)a->ndecay;
+    theta = 0.0;
+    for (i = 0; i <= a->ndecay; i++)
+    {
+        a->cdecay[i] = a->low_gain + (1.0 - a->low_gain) * 0.5 * (1.0 + cos (theta));
+        theta += delta;
+    }
+}
+
+void calc_buffs (DEXP a)
+{
+    a->trigsig   = (double *)malloc0 (2 * a->size * sizeof(complex));   // allow for double-sized output of filter
+    a->delsig    = (double *)malloc0 (    a->size * sizeof(complex));
+    a->audbuffer = (double *)malloc0 (    a->size * sizeof(complex));
+}
+
+void decalc_buffs (DEXP a)
+{
+    _aligned_free (a->audbuffer);
+    _aligned_free (a->delsig);
+    _aligned_free (a->trigsig);
+}
+
+void calc_dexp (DEXP a)
+{
+    // trigger signal preparation
+    a->avm = exp(-1.0 / (a->rate * a->dettau));
+    a->onem_avm = 1.0 - a->avm;
+    a->avsig = 0.0;
+    // level change
+    a->nattack = (int)(a->tattack * a->rate);
+    a->ndecay = (int)(a->tdecay * a->rate);
+    a->cattack = (double *)malloc0((a->nattack + 1) * sizeof(double));
+    a->cdecay = (double *)malloc0((a->ndecay + 1) * sizeof(double));
+    a->low_gain = 1.0 / a->exp_ratio;
+    calc_slews(a);
+    // control
+    a->state = 0;
+    a->count = 0;
+    a->hold_thresh = a->hysteresis_ratio * a->attack_thresh;    // hysteresis ratio < 1.0
+    a->nhold = (int)(a->thold * a->rate);
+    // vox
+    a->vox_count = (int)(a->audelay * a->rate);
+    // audio delay
+    a->audring = calc_delring ((int)a->rate, a->size, (int)(a->audelay * a->rate), a->audbuffer, a->out);
+}
+
+void decalc_dexp (DEXP a)
+{
+    decalc_delring (a->audring);
+    _aligned_free (a->cdecay);
+    _aligned_free (a->cattack);
+}
+
+void calc_filter (DEXP a)
+{
+    double* impulse;
+    // 2.0 gain on filter is somewhat arbitrarily chosen to get trigger input similar to that without the filter, knowing
+    //    that for any reasonable use of the filter there will be a reduction in trigger signal.
+    impulse = fir_bandpass (a->nc, a->low_cut, a->high_cut, a->rate, a->wintype, 1, 2.0/(double)(2 * a->size));
+    // print_impulse ("scf.txt", a->nc, impulse, 1, 0);
+    a->p = create_fircore (a->size, a->in, a->trigsig, a->nc, 1, impulse);
+    _aligned_free (impulse);
+    a->scdring = calc_delring (a->size + a->nc / 2, a->size, a->nc / 64, a->in, a->delsig);
+}
+
+void decalc_filter (DEXP a)
+{
+    destroy_fircore (a->p);
+    decalc_delring (a->scdring);
+}
+
+void calc_antivox(DEXP a)
+{
+    a->antivox_mult = exp(-1.0 / (a->antivox_rate * a->antivox_tau));
+    a->antivox_onemmult = 1.0 - a->antivox_mult;
+    a->antivox_data = (double *) malloc0 (a->antivox_size * sizeof (complex));
+}
+
+void decalc_antivox(DEXP a)
+{
+    _aligned_free (a->antivox_data);
+}
+
+PORT
+void create_dexp (int id, int run_dexp, int size, double* in, double* out, int rate, double dettau, double tattack, double tdecay,
+    double thold, double exp_ratio, double hyst_ratio, double attack_thresh, int nc, int wtype, double lowcut, double highcut,
+    int run_filt, int run_vox, int run_audelay, double audelay, void (__stdcall *pushvox)(int id, int active),
+    int antivox_run, int antivox_size, int antivox_rate, double antivox_gain, double antivox_tau)
+{
+    DEXP a = (DEXP) malloc0 (sizeof (dexp));
+    a->id = id;
+    a->run_dexp = run_dexp;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = (double)rate;
+    a->dettau = dettau;
+    a->tattack = tattack;
+    a->tdecay = tdecay;
+    a->thold = thold;
+    a->exp_ratio = exp_ratio;
+    a->hysteresis_ratio = hyst_ratio;
+    a->attack_thresh = attack_thresh;
+    a->nc = nc;
+    a->wintype = wtype;
+    a->low_cut = lowcut;
+    a->high_cut = highcut;
+    a->run_filt = run_filt;
+    a->run_vox = run_vox;
+    a->run_audelay = run_audelay;
+    a->audelay = audelay;
+    a->pushvox = pushvox;
+    a->antivox_run = antivox_run;
+    a->antivox_size = antivox_size;
+    a->antivox_rate = (double)antivox_rate;
+    a->antivox_gain = antivox_gain;
+    a->antivox_tau = antivox_tau;
+    calc_buffs (a);
+    calc_dexp (a);
+    calc_filter (a);
+    calc_antivox (a);
+    InitializeCriticalSectionAndSpinCount(&a->cs_update, 2500);
+    pdexp[id] = a;
+    return;
+}
+
+PORT
+void destroy_dexp (int id)
+{
+    DEXP a = pdexp[id];
+    DeleteCriticalSection (&a->cs_update);
+    decalc_antivox (a);
+    decalc_filter (a);
+    decalc_dexp (a);
+    decalc_buffs (a);
+    _aligned_free (a);
+}
+
+PORT
+void flush_dexp (int id)
+{
+    DEXP a = pdexp[id];
+    memset (a->audbuffer, 0, a->size * sizeof (complex));
+    memset (a->trigsig, 0, a->size * sizeof (complex));
+    memset (a->delsig,  0, a->size * sizeof (complex));
+    a->avsig = 0.0;
+    a->state = 0;
+    a->count = 0;
+    flush_fircore (a->p);
+    flush_delring (a->scdring);
+    flush_delring (a->audring);
+}
+
+enum _dexpstate
+{
+    DEXP_LOW,
+    DEXP_ATTACK,
+    DEXP_HIGH,
+    DEXP_HOLD,
+    DEXP_DECAY
+};
+
+PORT
+void xdexp (int id)
+{
+    DEXP a = pdexp[id];
+    int i;
+    double sig, gain, asig;
+    double max = 0.0;
+    EnterCriticalSection (&a->cs_update);
+
+    // ******* BEGIN SIDE-CHANNEL FILTER *******
+    if (a->run_filt)
+    {
+        xdelring (a->scdring);      // input is 'a->in'; output is 'a->delsig'
+        xfircore (a->p);            // input is 'a->in'; output is 'a->trigsig'
+    }
+    else
+    {
+        memcpy (a->delsig,  a->in, a->size * sizeof (complex));
+        memcpy (a->trigsig, a->in, a->size * sizeof (complex));
+    }
+    // ******* END SIDE-CHANNEL FILTER *******
+
+    // ******* CALCULATE ANTIVOX LEVEL *******
+    if (a->state == DEXP_LOW && a->antivox_new != 0)
+    {
+        // if VOX is currently NOT triggered, and, if we have new antivox data to process
+        for (i = 0; i < a->antivox_size; i++)
+        {
+            sig = sqrt (a->antivox_data[2 * i + 0] * a->antivox_data[2 * i + 0] + a->antivox_data[2 * i + 1] * a->antivox_data[2 * i + 1]);
+            a->antivox_level = a->antivox_mult * a->antivox_level + a->antivox_onemmult * sig;
+        }
+        // set the new_data flag to zero
+        a->antivox_new = 0;
+    }
+    // ******* END CALCULATE ANTIVOX LEVEL *******
+
+    // ******* BEGIN DEXP *******
+    // uses 'a->trigsig' as trigger signal; uses 'a->delsig' as audio input
+    // 'a->audbuffer' is audio output
+    // DEXP code runs continuously so it can be used to trigger VOX also.
+    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;
+        if (a->avsig > max)  max = a->avsig;
+        switch (a->state)
+        {
+        case DEXP_LOW:
+            if (a->antivox_run)
+                asig = a->avsig - a->antivox_gain * a->antivox_level;
+            else
+                asig = a->avsig;
+            if (asig > a->attack_thresh)
+            {
+                a->state = DEXP_ATTACK;
+                a->count = a->nattack;
+            }
+            a->audbuffer[2 * i + 0] = a->low_gain * a->delsig[2 * i + 0];
+            a->audbuffer[2 * i + 1] = a->low_gain * a->delsig[2 * i + 1];
+
+            // ******* BEGIN VOX *******
+            // If we're going to attack, turn on VOX immediately.
+            // Prepare 'vox_count' for the next turnoff too.
+            if (a->run_vox && a->state == DEXP_ATTACK)
+            {
+                (a->pushvox)(a->id, 1);
+                // Set vox_count for delay IF the audio delay is also enabled.
+                if (a->run_audelay)
+                    a->vox_count = (int)(a->audelay * a->rate);
+                else
+                    a->vox_count = 1;
+            }
+            // If we're sitting in this state and the delayed vox count expires, turn OFF VOX.
+            else if (a->run_vox && --(a->vox_count) == 0)
+                (a->pushvox)(a->id, 0);
+            // Don't let 'vox_count' keep counting down.
+            else if (a->vox_count < 0)
+                a->vox_count = 0;
+            // ******* END VOX *******
+
+            break;
+        case DEXP_ATTACK:
+            gain = a->cattack[a->nattack - a->count];
+            a->audbuffer[2 * i + 0] = a->delsig[2 * i + 0] * gain;
+            a->audbuffer[2 * i + 1] = a->delsig[2 * i + 1] * gain;
+            if (a->count-- == 0)
+                a->state = DEXP_HIGH;
+            break;
+        case DEXP_HIGH:
+            if (a->avsig < a->hold_thresh)
+            {
+                a->state = DEXP_HOLD;
+                a->count = a->nhold;
+            }
+            a->audbuffer[2 * i + 0] = a->delsig[2 * i + 0];
+            a->audbuffer[2 * i + 1] = a->delsig[2 * i + 1];
+            break;
+        case DEXP_HOLD:
+            a->audbuffer[2 * i + 0] = a->delsig[2 * i + 0];
+            a->audbuffer[2 * i + 1] = a->delsig[2 * i + 1];
+            if (a->avsig > a->attack_thresh)
+                a->state = DEXP_HIGH;
+            else if (a->count-- == 0)
+            {
+                a->state = DEXP_DECAY;
+                a->count = a->ndecay;
+            }
+            break;
+        case DEXP_DECAY:
+            gain = a->cdecay[a->ndecay - a->count];
+            a->audbuffer[2 * i + 0] = a->delsig[2 * i + 0] * gain;
+            a->audbuffer[2 * i + 1] = a->delsig[2 * i + 1] * gain;
+            if (a->count-- == 0)
+                a->state = DEXP_LOW;
+            break;
+        }
+    }
+    a->peak = max;
+    // If DEXP functionality is set to OFF, copy its input to overwrite its output.
+    if (!a->run_dexp)
+        memcpy (a->audbuffer, a->delsig, a->size * sizeof (complex));
+    // ******* END DEXP *******
+
+    // ******* BEGIN AUDIO DELAY *******
+    if (a->run_audelay)
+        xdelring (a->audring);      // uses 'a->audbuffer' as audio input; uses 'a->out' as audio output
+    else
+        memcpy (a->out, a->audbuffer, a->size * sizeof (complex));
+    // ******* END AUDIO DELAY *******
+
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SendCBPushDexpVox (int id, void (__stdcall *pushvox)(int id, int active))
+{
+    // Call to set the address of the callback to operate VOX.
+    DEXP a = pdexp[id];
+    a->pushvox = pushvox;
+}
+
+PORT
+void SetDEXPRun (int id, int run)
+{
+    // run != 0, puts dexp in the audio processing path; otherwise, it's only used to trigger VOX
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run_dexp = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPSize (int id, int size)
+{
+    // There are some constraints on the input/output buffer sizes.
+    //   * must be a power-of-two
+    //   * must be less than or equal to 'nc', the number of filter coefficients, which is also a power-of-two
+    //   * must be less than 'rate' samples because of the sizing of 'audring'
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    decalc_dexp (a);
+    decalc_buffs (a);
+    a->size = size;
+    calc_buffs (a);
+    calc_dexp (a);
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPIOBuffers (int id, double* in, double* out)
+{
+    // Sets the input/output buffers.  They can be the same.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    decalc_dexp (a);
+    a->in = in;
+    a->out = out;
+    calc_dexp (a);
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPRate (int id, double rate)
+{
+    // Sets the sample rate.
+    // This is used for timing and filter calculations as well as sizing 'audring'.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    decalc_dexp (a);
+    a->rate = rate;
+    calc_dexp (a);
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPDetectorTau (int id, double tau)
+{
+    // Time-constant for smoothing the signal for detection (seconds).
+    // 0.01 seconds is a good starting point to try.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->dettau = tau;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPAttackTime (int id, double time)
+{
+    // Set attack time, seconds.
+    // 0.002 - 0.100 should be a good range.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->tattack = time;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPReleaseTime (int id, double time)
+{
+    // Set release time, seconds.
+    // 0.002 - 0.999 should be a good range.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->tdecay = time;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPHoldTime (int id, double time)
+{
+    // Set hold time, seconds.
+    // 0.000 - 2.000 should be a good range.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->thold = time;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPExpansionRatio (int id, double ratio)
+{
+    // Set expansion ratio.  High_gain = 1.0; Low_gain = 1.0/exp_ratio.
+    // Range of 1.0 - 30.0 should be good.  Could use dB:  0.0 - 30.0dB.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->exp_ratio = ratio;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPHysteresisRatio (int id, double ratio)
+{
+    // Set Hysteresis Ratio.  Hold_thresh = hysteresis_ratio * Attack_thresh.
+    // Expose to operator in dB:  0.0dB - 9.9dB should be good (1.000 - 0.320).
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->hysteresis_ratio = ratio;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPAttackThreshold (int id, double thresh)
+{
+    // Set attack threshold.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->attack_thresh = thresh;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPFilterTaps (int id, int taps)
+{
+    // Set number of taps.  Must be a power of two and an even multiple of 'size'.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    a->nc = taps;
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPWindowType (int id, int type)
+{
+    // Set filter window type.
+    //   * 0 - 4-term Blackman-Harris.
+    //   * 1 - 7-term Blackman-Harris.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    a->wintype = type;
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPLowCut (int id, double lowcut)
+{
+    // Set side-channel filter low_cut (Hertz).
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    a->low_cut = lowcut;
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPHighCut (int id, double highcut)
+{
+    // Set side-channel filter high_cut (Hertz).
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_filter (a);
+    a->high_cut = highcut;
+    calc_filter (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPRunSideChannelFilter (int id, int run)
+{
+    // Turn OFF/ON the side-channel filter and its compensating delay.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run_filt = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPRunVox (int id, int run)
+{
+    // Turn OFF/ON calls to 'pushvox(...)'.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run_vox = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPRunAudioDelay (int id, int run)
+{
+    // Turn OFF/ON audio delay line.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run_audelay = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetDEXPAudioDelay (int id, double delay)
+{
+    // Set the audio delay, seconds.
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_dexp (a);
+    a->audelay = delay;
+    calc_dexp (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void GetDEXPPeakSignal (int id, double* peak)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    *peak = a->peak;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetAntiVOXRun (int id, int run)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->antivox_run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetAntiVOXSize (int id, int size)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_antivox(a);
+    a->antivox_size = size;
+    calc_antivox(a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetAntiVOXRate (int id, double rate)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_antivox(a);
+    a->antivox_rate = rate;
+    calc_antivox(a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetAntiVOXGain (int id, double gain)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    a->antivox_gain = gain;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetAntiVOXDetectorTau (int id, double tau)
+{
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    decalc_antivox (a);
+    a->antivox_tau = tau;
+    calc_antivox (a);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SendAntiVOXData (int id, int nsamples, double* data)
+{
+    // note:  'nsamples' is not used as it has been previously specified
+    DEXP a = pdexp[id];
+    EnterCriticalSection (&a->cs_update);
+    memcpy (a->antivox_data, data, a->antivox_size * sizeof (complex));
+    a->antivox_new = 1;
+    LeaveCriticalSection (&a->cs_update);
+}

wdsp/dexp.h

@@ -0,0 +1,123 @@
+/*  dexp.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2018, 2019 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 _dexp_h
+#define _dexp_h
+
+typedef struct _delring
+{
+    int rsize;                          // ringsize (measured in complex samples)
+    double* ring;                       // ring buffer
+    int inptr;                          // ring input pointer (counts in complex samples)
+    int outptr;                         // ring output pointer (counts in complex samples)
+    int rdelay;                         // ring delay (measured in complex samples)
+    int size;                           // input/output size in complex samples
+    double* in;                         // source buffer
+    double* out;                        // destination buffer
+} delring, *DELRING;
+
+typedef struct _dexp
+{
+    int id;                             // 'id' for this dexp
+    int run_dexp;                       // 0 if dexp is OFF; 1 if it's ON
+    int size;                           // size of input/output buffers
+    double* in;                         // audio input buffer
+    double* out;                        // audio output buffer; can be same as 'in'
+    double rate;                        // sample rate
+    double dettau;                      // detection averaging time constant
+    double avm;                         // averaging multiplier
+    double onem_avm;                    // one minus averaging multiplier
+    double avsig;                       // averaged detection signal
+    int state;                          // state machine control
+    int count;                          // count variable used within a state
+    double tattack;                     // attack time
+    double tdecay;                      // decay time
+    int nattack;                        // one less than total number of attack multipliers
+    int ndecay;                         // one less than total number of decay multipliers
+    double* cattack;                    // attack curve multipliers
+    double* cdecay;                     // decay curve multipliers
+    double attack_thresh;               // attack threshold
+    double hold_thresh;                 // hold & decay threshold
+    double thold;                       // hold time
+    int nhold;                          // hold count
+    double exp_ratio;                   // expander ratio (high-gain to low-gain)
+    double hysteresis_ratio;            // ratio hold_thresh/attack_thresh.  0.0 < ratio < 1.0
+    double low_gain;                    // gain when gate is closed
+    double* trigsig;                    // buffer for trigger signal (signal after side-channel filter)
+    double* delsig;                     // buffer for signal delayed to match trigger signal
+    double peak;                        // peak signal value to return to console
+    // side-channel bandpass filter & and buffer for compensating delay
+    int run_filt;                       // 1 = side-channel filter and compensating delay are ON, 0 = OFF
+    int nc;                             // number of coefficients
+    int wintype;                        // window type
+    double low_cut;                     // low cutoff frequency
+    double high_cut;                    // high cutoff frequency
+    FIRCORE p;                          // filter structure
+    DELRING scdring;                    // delay ring for side channel
+    // output audio delay to cover RF_Delay + Xmtr_delay_and_upslew
+    double* audbuffer;                  // buffer to serve as input to audring
+    int run_audelay;                    // 'run' variable for audio delay ring
+    double audelay;                     // audio output delay in seconds
+    DELRING audring;                    // audio delay ring
+    // vox
+    int run_vox;
+    void (__stdcall *pushvox)(int channel, int active);
+    int vox_count;
+    // update critical section
+    CRITICAL_SECTION cs_update;
+    // anti-vox
+    int antivox_run;                    // 'run' for anti-vox
+    int antivox_new;                    // internal variable indicating new anti-vox data is available
+    int antivox_size;                   // size of anti-vox data buffer
+    double antivox_rate;                // sample-rate of anti-vox data
+    double antivox_tau;                 // time-constant of anti-vox smoothing
+    double antivox_gain;                // anti-vox gain factor
+    double antivox_mult;                // multiplier for anti-vox smoothing
+    double antivox_onemmult;            // one minus antivox_mult
+    double antivox_level;               // current anti-vox smoothed signal level
+    double* antivox_data;               // buffer to hold new anti-vox data
+} dexp, *DEXP;
+
+extern DEXP pdexp[];
+
+__declspec (dllexport) void create_dexp (int id, int run_dexp, int size, double* in, double* out, int rate, double dettau, double tattack, double tdecay,
+    double thold, double exp_ratio, double hyst_ratio, double attack_thresh, int nc, int wtype, double lowcut, double highcut,
+    int run_filt, int run_vox, int run_audelay, double audelay, void (__stdcall *pushvox)(int id, int active),
+    int antivox_run, int antivox_size, int antivox_rate, double antivox_gain, double antivox_tau);
+
+__declspec (dllexport) void destroy_dexp (int id);
+
+__declspec (dllexport) void flush_dexp (int id);
+
+__declspec (dllexport) void xdexp (int id);
+
+__declspec (dllexport) void SetDEXPSize (int id, int size);
+
+__declspec (dllexport) void SetDEXPRate (int id, double rate);
+
+__declspec (dllexport) void SendAntiVOXData (int id, int nsamples, double* data);
+
+#endif

wdsp/div.c

@@ -0,0 +1,219 @@
+/*  div.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014 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"
+
+#define MAX_NR  (8)     // maximum number of receivers to mix
+
+MDIV create_div (int run, int nr, int size, double **in, double *out)
+{
+    int i;
+    MDIV a = (MDIV) malloc0 (sizeof (mdiv));
+    a->run = run;
+    a->nr = nr;
+    a->size = size;
+    a->out = out;
+    a->in = (double **) malloc0 ( MAX_NR * sizeof (double *));
+    if (in != 0)
+        for (i = 0; i < nr; i++) a->in[i] = in[i];
+    a->Irotate = (double *) malloc0 (MAX_NR * sizeof (double));
+    a->Qrotate = (double *) malloc0 (MAX_NR * sizeof (double));
+    InitializeCriticalSectionAndSpinCount (&a->cs_update, 2500);
+    for (i = 0; i < 4; i++)                                                                                 ///////////// legacy interface - remove
+        a->legacy[i] = (double *) malloc0 (2048 * sizeof (complex));                                        ///////////// legacy interface - remove
+    return a;
+}
+
+void destroy_div (MDIV a)
+{
+    int i;                                                                                                  ///////////// legacy interface - remove
+    DeleteCriticalSection (&a->cs_update);
+    for (i = 0; i < 4; i++)                                                                                 ///////////// legacy interface - remove
+        _aligned_free (a->legacy[i]);                                                                       ///////////// legacy interface - remove
+    _aligned_free (a->Qrotate);
+    _aligned_free (a->Irotate);
+    _aligned_free (a->in);
+    _aligned_free (a);
+}
+
+void flush_div (MDIV a)
+{
+
+}
+
+void xdiv (MDIV a)
+{
+    if (a->run)
+    {
+        EnterCriticalSection (&a->cs_update);
+        if (a->output != a->nr)
+        {
+            if (a->out != a->in[a->output])
+                memcpy (a->out, a->in[a->output], a->size * sizeof (complex));
+        }
+        else
+        {
+            int i, j;
+            double I, Q;
+            memset (a->out, 0, a->size * sizeof (complex));
+            for (i = 0; i < a->nr; i++)
+                for (j = 0; j < a->size; j++)
+                {
+                    I = a->in[i][2 * j + 0];
+                    Q = a->in[i][2 * j + 1];
+                    a->out[2 * j + 0] += a->Irotate[i] * I - a->Qrotate[i] * Q;
+                    a->out[2 * j + 1] += a->Irotate[i] * Q + a->Qrotate[i] * I;
+                }
+        }
+        LeaveCriticalSection (&a->cs_update);
+    }
+    else
+        memcpy (a->out, a->in[0], a->size * sizeof (complex));
+}
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       CALLS FOR EXTERNAL USE                                          *
+*                                                                                                       *
+********************************************************************************************************/
+
+#define MAX_EXT_DIVS    (2)                         // maximum number of DIVs called from outside wdsp
+__declspec (align (16)) MDIV pdiv[MAX_EXT_DIVS];    // array of pointers for DIVs used EXTERNAL to wdsp
+
+PORT
+void create_divEXT (int id, int run, int nr, int size)
+{
+    pdiv[id] = create_div (run, nr, size, 0, 0);
+}
+
+PORT
+void destroy_divEXT (int id)
+{
+    destroy_div (pdiv[id]);
+}
+
+PORT
+void flush_divEXT (int id)
+{
+    flush_div (pdiv[id]);
+}
+
+PORT
+void xdivEXT (int id, int nsamples, double **in, double *out)
+{
+    int i;
+    MDIV a = pdiv[id];
+    a->size = nsamples;
+    a->out = out;
+    for (i = 0; i < a->nr; i++) a->in[i] = in[i];
+    xdiv (a);
+}
+
+// 0 - does nothing; 1 - operates
+PORT
+void SetEXTDIVRun (int id, int run)
+{
+    MDIV a = pdiv[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+// size of data buffer in complex samples
+PORT
+void SetEXTDIVBuffsize (int id, int size)
+{
+    MDIV a = pdiv[id];
+    EnterCriticalSection (&a->cs_update);
+    a->size = size;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+// number of receivers being used for diversity
+PORT
+void SetEXTDIVNr (int id, int nr)
+{
+    MDIV a = pdiv[id];
+    EnterCriticalSection (&a->cs_update);
+    a->nr = nr;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+// number of which receiver to output
+//  if output==nr, mixing occurs
+PORT
+void SetEXTDIVOutput (int id, int output)
+{
+    MDIV a = pdiv[id];
+    EnterCriticalSection (&a->cs_update);
+    a->output = output;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+// I and Q "rotate" multipliers for each receiver
+//  can be set to 1.0 and 0.0 for "reference receiver"
+PORT
+void SetEXTDIVRotate (int id, int nr, double *Irotate, double *Qrotate)
+{
+    MDIV a = pdiv[id];
+    EnterCriticalSection (&a->cs_update);
+    memcpy (a->Irotate, Irotate, nr * sizeof (double));
+    memcpy (a->Qrotate, Qrotate, nr * sizeof (double));
+    LeaveCriticalSection (&a->cs_update);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                     LEGACY INTERFACE - REMOVE                                         *
+*                                                                                                       *
+********************************************************************************************************/
+PORT
+void xdivEXTF (int id, int size, float **input, float *Iout, float *Qout)
+{
+    int i, j;
+    MDIV a = pdiv[id];
+    if (a->run)
+    {
+        a->size = size;
+        for (i = 0; i < a->nr; i++)
+        {
+            for (j = 0; j < a->size; j++)
+            {
+                a->legacy[i][2 * j + 0] = (double)input[2 * i + 0][j];
+                a->legacy[i][2 * j + 1] = (double)input[2 * i + 1][j];
+            }
+            a->in[i] = a->legacy[i];
+        }
+        a->out = a->legacy[3];
+        xdiv (a);
+        for (j = 0; j < a->size; j++)
+        {
+            Iout[j] = (float)a->legacy[3][2 * j + 0];
+            Qout[j] = (float)a->legacy[3][2 * j + 1];
+        }
+    }
+}

wdsp/div.h

@@ -0,0 +1,56 @@
+/*  div.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014 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 _div_h
+#define _div_h
+
+typedef struct _div
+{
+    int run;
+    int nr;                         // number of receivers to mix
+    int size;                       // size of input/output buffers
+    double **in;                    // input buffers
+    double *out;                    // output buffer
+    int output;                     // which rcvr to output; ==nr for mix
+    double *Irotate;
+    double *Qrotate;
+    CRITICAL_SECTION cs_update;
+    double *legacy[4];                                                                  ///////////// legacy interface - remove
+} mdiv, *MDIV;
+
+extern MDIV create_div (int run, int nr, int size, double **in, double *out);
+
+extern void destroy_div (MDIV pdiv);
+
+extern void xdiv (MDIV pdiv);
+
+extern __declspec(dllexport) void xdivEXT (int id, int nsamples, double **in, double *out);
+
+extern __declspec(dllexport) void create_divEXT (int id, int run, int nr, int size);
+
+extern __declspec(dllexport) void destroy_divEXT (int id);
+
+#endif

wdsp/eer.c

@@ -0,0 +1,393 @@
+/*  eer.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014 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"
+
+PORT
+EER create_eer (int run, int size, double* in, double* out, double* outM, int rate, double mgain, double pgain, int rundelays, double mdelay, double pdelay, int amiq)
+{
+    EER a = (EER) malloc0 (sizeof (eer));
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->outM = outM;
+    a->rate = rate;
+    a->mgain = mgain;
+    a->pgain = pgain;
+    a->rundelays = rundelays;
+    a->mdelay = mdelay;
+    a->pdelay = pdelay;
+    a->amiq = amiq;
+    a->mdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->outM,                                    // input buffer
+        a->outM,                                    // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->mdelay);                                 // delay
+    a->pdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->out,                                     // input buffer
+        a->out,                                     // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->pdelay);                                 // delay
+    InitializeCriticalSectionAndSpinCount(&a->cs_update, 2500);
+
+    a->legacy  = (double *) malloc0 (2048 * sizeof (complex));                                                      /////////////// legacy interface - remove
+    a->legacyM = (double *) malloc0 (2048 * sizeof (complex));                                                      /////////////// legacy interface - remove
+
+    return a;
+}
+
+PORT
+void destroy_eer (EER a)
+{
+    DeleteCriticalSection (&a->cs_update);
+    destroy_delay (a->pdel);
+    destroy_delay (a->mdel);
+    _aligned_free (a);
+}
+
+PORT
+void flush_eer (EER a)
+{
+    flush_delay (a->mdel);
+    flush_delay (a->pdel);
+}
+
+PORT
+void xeer (EER a)
+{
+    EnterCriticalSection (&a->cs_update);
+    if (a->run)
+    {
+        int i;
+        double I, Q, mag;
+        for (i = 0; i < a->size; i++)
+        {
+            I = a->in[2 * i + 0];
+            Q = a->in[2 * i + 1];
+            a->outM[2 * i + 0] = I * a->mgain;
+            a->outM[2 * i + 1] = Q * a->mgain;
+            switch (a->amiq)
+            {
+            case 0:     // send phase info only, magnitude is constant
+                mag = sqrt (I * I + Q * Q);
+                a->out [2 * i + 0] = a->pgain * I / mag;
+                a->out [2 * i + 1] = a->pgain * Q / mag;
+                break;
+            case 1:     // send magnitude and phase information, I and Q
+                a->out [2 * i + 0] = a->pgain * I;
+                a->out [2 * i + 1] = a->pgain * Q;
+                break;
+            case 2:     // send envelope
+                mag = sqrt (I * I + Q * Q);
+                a->out [2 * i + 0] = a->out[2 * i + 1] = a->pgain * mag;
+                break;
+            }
+        }
+        xdelay (a->mdel);       // delay for outM
+        xdelay (a->pdel);       // delay for out
+    }
+    else if (a->out != a->in)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+    LeaveCriticalSection (&a->cs_update);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       CALLS FOR EXTERNAL USE                                          *
+*                                                                                                       *
+********************************************************************************************************/
+
+#define MAX_EXT_EERS    (2)                         // maximum number of EERs called from outside wdsp
+__declspec (align (16)) EER peer[MAX_EXT_EERS];     // array of pointers for EERs used EXTERNAL to wdsp
+
+
+PORT
+void create_eerEXT (int id, int run, int size, int rate, double mgain, double pgain, int rundelays, double mdelay, double pdelay, int amiq)
+{
+    peer[id] = create_eer (run, size, 0, 0, 0, rate, mgain, pgain, rundelays, mdelay, pdelay, amiq);
+}
+
+PORT
+void destroy_eerEXT (int id)
+{
+    destroy_eer (peer[id]);
+}
+
+PORT
+void flush_eerEXT (int id)
+{
+    flush_eer (peer[id]);
+}
+
+PORT
+void SetEERRun (int id, int run)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERAMIQ (int id, int amiq)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->amiq = amiq;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERMgain (int id, double gain)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->mgain = gain;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERPgain (int id, double gain)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->pgain = gain;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERRunDelays (int id, int run)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->rundelays = run;
+    SetDelayRun (a->mdel, a->rundelays);
+    SetDelayRun (a->pdel, a->rundelays);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERMdelay (int id, double delay)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->mdelay = delay;
+    SetDelayValue (a->mdel, a->mdelay);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERPdelay (int id, double delay)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->pdelay = delay;
+    SetDelayValue (a->pdel, a->pdelay);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERSize (int id, int size)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->size = size;
+    SetDelayBuffs (a->mdel, a->size, a->outM, a->outM);
+    SetDelayBuffs (a->pdel, a->size, a->out, a->out);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void SetEERSamplerate (int id, int rate)
+{
+    EER a = peer[id];
+    EnterCriticalSection (&a->cs_update);
+    a->rate = rate;
+    destroy_delay (a->mdel);
+    destroy_delay (a->pdel);
+    a->mdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->outM,                                    // input buffer
+        a->outM,                                    // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->mdelay);                                 // delay
+    a->pdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->out,                                     // input buffer
+        a->out,                                     // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->pdelay);                                 // delay
+    LeaveCriticalSection (&a->cs_update);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           POINTER-BASED PROPERTIES                                    *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void pSetEERRun (EER a, int run)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERAMIQ (EER a, int amiq)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->amiq = amiq;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERMgain (EER a, double gain)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->mgain = gain;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERPgain (EER a, double gain)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->pgain = gain;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERRunDelays (EER a, int run)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->rundelays = run;
+    SetDelayRun (a->mdel, a->rundelays);
+    SetDelayRun (a->pdel, a->rundelays);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERMdelay (EER a, double delay)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->mdelay = delay;
+    SetDelayValue (a->mdel, a->mdelay);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERPdelay (EER a, double delay)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->pdelay = delay;
+    SetDelayValue (a->pdel, a->pdelay);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERSize (EER a, int size)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->size = size;
+    SetDelayBuffs (a->mdel, a->size, a->outM, a->outM);
+    SetDelayBuffs (a->pdel, a->size, a->out, a->out);
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetEERSamplerate (EER a, int rate)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->rate = rate;
+    destroy_delay (a->mdel);
+    destroy_delay (a->pdel);
+    a->mdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->outM,                                    // input buffer
+        a->outM,                                    // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->mdelay);                                 // delay
+    a->pdel = create_delay (
+        a->rundelays,                               // run
+        a->size,                                    // size
+        a->out,                                     // input buffer
+        a->out,                                     // output buffer
+        a->rate,                                    // sample rate
+        20.0e-09,                                   // delta (delay stepsize)
+        a->pdelay);                                 // delay
+    LeaveCriticalSection (&a->cs_update);
+}
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                          Legacy Interface                                             *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void xeerEXTF (int id, float* inI, float* inQ, float* outI, float* outQ, float* outMI, float* outMQ, int mox, int size)
+{
+    EER a = peer[id];
+    if (mox && a->run)
+    {
+        int i;
+        a->in   = a->legacy;
+        a->out  = a->legacy;
+        a->outM = a->legacyM;
+        a->size = size;
+        SetDelayBuffs (a->mdel, a->size, a->outM, a->outM);
+        SetDelayBuffs (a->pdel, a->size, a->out, a->out);
+        for (i = 0; i < a->size; i++)
+        {
+            a->legacy[2 * i + 0] = (double)inI[i];
+            a->legacy[2 * i + 1] = (double)inQ[i];
+        }
+        xeer (a);
+        for (i = 0; i < a->size; i++)
+        {
+            outI[i]  = (float)a->legacy [2 * i + 0];
+            outQ[i]  = (float)a->legacy [2 * i + 1];
+            outMI[i] = (float)a->legacyM[2 * i + 0];
+            outMQ[i] = (float)a->legacyM[2 * i + 1];
+        }
+    }
+}

wdsp/eer.h

@@ -0,0 +1,77 @@
+/*  eer.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014 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 _eer_h
+#define _eer_h
+
+typedef struct _eer
+{
+    int run;
+    int amiq;
+    int size;
+    double* in;
+    double* out;
+    double* outM;
+    int rate;
+    double mgain;
+    double pgain;
+    int rundelays;
+    double mdelay;
+    double pdelay;
+    DELAY mdel;
+    DELAY pdel;
+    CRITICAL_SECTION cs_update;
+    double *legacy;                                                                                                     ////////////  legacy interface - remove
+    double *legacyM;                                                                                                    ////////////  legacy interface - remove
+} eer, *EER;
+
+__declspec (dllexport) EER create_eer (int run, int size, double* in, double* out, double* outM, int rate, double mgain, double pgain, int rundelays, double mdelay, double pdelay, int amiq);
+
+__declspec (dllexport) void destroy_eer (EER a);
+
+__declspec (dllexport) void flush_eer (EER a);
+
+__declspec (dllexport) void xeer (EER a);
+
+__declspec (dllexport) void pSetEERRun (EER a, int run);
+
+__declspec (dllexport) void pSetEERAMIQ (EER a, int amiq);
+
+__declspec (dllexport) void pSetEERMgain (EER a, double gain);
+
+__declspec (dllexport) void pSetEERPgain (EER a, double gain);
+
+__declspec (dllexport) void pSetEERRunDelays (EER a, int run);
+
+__declspec (dllexport) void pSetEERMdelay (EER a, double delay);
+
+__declspec (dllexport) void pSetEERPdelay (EER a, double delay);
+
+__declspec (dllexport) void pSetEERSize (EER a, int size);
+
+__declspec (dllexport) void pSetEERSamplerate (EER a, int rate);
+
+#endif

wdsp/emnr.h

@@ -0,0 +1,207 @@
+/*  emnr.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2015 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 _emnr_h
+#define _emnr_h
+
+typedef struct _emnr
+{
+    int run;
+    int position;
+    int bsize;
+    double* in;
+    double* out;
+    int fsize;
+    int ovrlp;
+    int incr;
+    double* window;
+    int iasize;
+    double* inaccum;
+    double* forfftin;
+    double* forfftout;
+    int msize;
+    double* mask;
+    double* revfftin;
+    double* revfftout;
+    double** save;
+    int oasize;
+    double* outaccum;
+    double rate;
+    int wintype;
+    double ogain;
+    double gain;
+    int nsamps;
+    int iainidx;
+    int iaoutidx;
+    int init_oainidx;
+    int oainidx;
+    int oaoutidx;
+    int saveidx;
+    fftw_plan Rfor;
+    fftw_plan Rrev;
+    struct _g
+    {
+        int gain_method;
+        int npe_method;
+        int ae_run;
+        double msize;
+        double* mask;
+        double* y;
+        double* lambda_y;
+        double* lambda_d;
+        double* prev_mask;
+        double* prev_gamma;
+        double gf1p5;
+        double alpha;
+        double eps_floor;
+        double gamma_max;
+        double xi_min;
+        double q;
+        double gmax;
+        //
+        double* GG;
+        double* GGS;
+        FILE* fileb;
+        //
+        int dim_zeta;
+        double* zeta_hat;
+        int* zeta_true;
+        double z_gamma_min;
+        double z_gamma_max;
+        double z_xihat_min;
+        double z_xihat_max;
+        double zeta_thresh;
+    } g;
+    struct _npest
+    {
+        int incr;
+        double rate;
+        int msize;
+        double* lambda_y;
+        double* lambda_d;
+        double* p;
+        double* alphaOptHat;
+        double alphaC;
+        double alphaCsmooth;
+        double alphaCmin;
+        double* alphaHat;
+        double alphaMax;
+        double* sigma2N;
+        double alphaMin_max_value;
+        double snrq;
+        double betamax;
+        double* pbar;
+        double* p2bar;
+        double invQeqMax;
+        double av;
+        double* Qeq;
+        int U;
+        double Dtime;
+        int V;
+        int D;
+        double MofD;
+        double MofV;
+        double* bmin;
+        double* bmin_sub;
+        int* k_mod;
+        double* actmin;
+        double* actmin_sub;
+        int subwc;
+        int* lmin_flag;
+        double* pmin_u;
+        double invQbar_points[4];
+        double nsmax[4];
+        double** actminbuff;
+        int amb_idx;
+    } np;
+    struct _npests
+    {
+        int incr;
+        double rate;
+        int msize;
+        double* lambda_y;
+        double* lambda_d;
+
+        double alpha_pow;
+        double alpha_Pbar;
+        double epsH1;
+        double epsH1r;
+
+        double* sigma2N;
+        double* PH1y;
+        double* Pbar;
+        double* EN2y;
+    } nps;
+    struct _npestl
+    {
+        double rate;
+        int msize;
+        int incr;
+        double* Ysq;
+        double* P;
+        double* Pmin;
+        double* p;
+        double* D;
+        double* lambda_d;
+
+        double eta;
+        double gamma;
+        double beta;
+        double delta_LF;
+        double delta_MF;
+        double delta_0;
+        double delta_1;
+        double delta_2;
+        double alpha_d;
+        double alpha_p;
+    } npl;
+    struct _ae
+    {
+        int msize;
+        double* lambda_y;
+        double zetaThresh;
+        double psi;
+        double* nmask;
+        double t2;
+    } ae;
+}emnr, *EMNR;
+
+extern EMNR create_emnr (int run, int position, int size, double* in, double* out, int fsize, int ovrlp,
+    int rate, int wintype, double gain, int gain_method, int npe_method, int ae_run);
+
+extern void destroy_emnr (EMNR a);
+
+extern void flush_emnr (EMNR a);
+
+extern void xemnr (EMNR a, int pos);
+
+extern void setBuffers_emnr (EMNR a, double* in, double* out);
+
+extern void setSamplerate_emnr (EMNR a, int rate);
+
+extern void setSize_emnr (EMNR a, int size);
+
+#endif

wdsp/emph.c

@@ -0,0 +1,270 @@
+/*  emph.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014, 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"
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Partitioned Overlap-Save FM Pre-Emphasis                                *
+*                                                                                                       *
+********************************************************************************************************/
+
+EMPHP create_emphp (int run, int position, int size, int nc, int mp, double* in, double* out, int rate, int ctype, double f_low, double f_high)
+{
+    EMPHP a = (EMPHP) malloc0 (sizeof (emphp));
+    double* impulse;
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->nc = nc;
+    a->mp = mp;
+    a->in = in;
+    a->out = out;
+    a->rate = rate;
+    a->ctype = ctype;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    impulse = fc_impulse (a->nc, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    a->p = create_fircore (a->size, a->in, a->out, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+    return a;
+}
+
+void destroy_emphp (EMPHP a)
+{
+    destroy_fircore (a->p);
+    _aligned_free (a);
+}
+
+void flush_emphp (EMPHP a)
+{
+    flush_fircore (a->p);
+}
+
+void xemphp (EMPHP a, int position)
+{
+    if (a->run && a->position == position)
+        xfircore (a->p);
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_emphp (EMPHP a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    setBuffers_fircore (a->p, a->in, a->out);
+}
+
+void setSamplerate_emphp (EMPHP a, int rate)
+{
+    double* impulse;
+    a->rate = rate;
+    impulse = fc_impulse (a->nc, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+void setSize_emphp (EMPHP a, int size)
+{
+    double* impulse;
+    a->size = size;
+    setSize_fircore (a->p, a->size);
+    impulse = fc_impulse (a->nc, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                       Partitioned Overlap-Save FM Pre-Emphasis:  TXA Properties                       *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXAFMEmphPosition (int channel, int position)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].preemph.p->position = position;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAFMEmphMP (int channel, int mp)
+{
+    EMPHP a;
+    a = txa[channel].preemph.p;
+    if (a->mp != mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}
+
+PORT
+void SetTXAFMEmphNC (int channel, int nc)
+{
+    EMPHP a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].preemph.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = fc_impulse (a->nc, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAFMPreEmphFreqs (int channel, double low, double high)
+{
+    EMPHP a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].preemph.p;
+    if (a->f_low != low || a->f_high != high)
+    {
+        a->f_low = low;
+        a->f_high = high;
+        impulse = fc_impulse (a->nc, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       Overlap-Save FM Pre-Emphasis                                    *
+*                                                                                                       *
+********************************************************************************************************/
+
+void calc_emph (EMPH a)
+{
+    a->infilt = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->product = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->mults = fc_mults(a->size, a->f_low, a->f_high, -20.0 * log10(a->f_high / a->f_low), 0.0, a->ctype, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    a->CFor = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->infilt, (fftw_complex *)a->product, FFTW_FORWARD, FFTW_PATIENT);
+    a->CRev = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->product, (fftw_complex *)a->out, FFTW_BACKWARD, FFTW_PATIENT);
+}
+
+void decalc_emph (EMPH a)
+{
+    fftw_destroy_plan(a->CRev);
+    fftw_destroy_plan(a->CFor);
+    _aligned_free(a->mults);
+    _aligned_free(a->product);
+    _aligned_free(a->infilt);
+}
+
+EMPH create_emph (int run, int position, int size, double* in, double* out, int rate, int ctype, double f_low, double f_high)
+{
+    EMPH a = (EMPH) malloc0 (sizeof (emph));
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = (double)rate;
+    a->ctype = ctype;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    calc_emph (a);
+    return a;
+}
+
+void destroy_emph (EMPH a)
+{
+    decalc_emph (a);
+    _aligned_free (a);
+}
+
+void flush_emph (EMPH a)
+{
+    memset (a->infilt, 0, 2 * a->size * sizeof (complex));
+}
+
+void xemph (EMPH a, int position)
+{
+    int i;
+    double I, Q;
+    if (a->run && a->position == position)
+    {
+        memcpy (&(a->infilt[2 * a->size]), a->in, a->size * sizeof (complex));
+        fftw_execute (a->CFor);
+        for (i = 0; i < 2 * a->size; i++)
+        {
+            I = a->product[2 * i + 0];
+            Q = a->product[2 * i + 1];
+            a->product[2 * i + 0] = I * a->mults[2 * i + 0] - Q * a->mults[2 * i + 1];
+            a->product[2 * i + 1] = I * a->mults[2 * i + 1] + Q * a->mults[2 * i + 0];
+        }
+        fftw_execute (a->CRev);
+        memcpy (a->infilt, &(a->infilt[2 * a->size]), a->size * sizeof(complex));
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_emph (EMPH a, double* in, double* out)
+{
+    decalc_emph (a);
+    a->in = in;
+    a->out = out;
+    calc_emph (a);
+}
+
+void setSamplerate_emph (EMPH a, int rate)
+{
+    decalc_emph (a);
+    a->rate = rate;
+    calc_emph (a);
+}
+
+void setSize_emph (EMPH a, int size)
+{
+    decalc_emph(a);
+    a->size = size;
+    calc_emph(a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Overlap-Save FM Pre-Emphasis:  TXA Properties                           *
+*                                                                                                       *
+********************************************************************************************************/
+/*  // Uncomment when needed
+PORT
+void SetTXAFMEmphPosition (int channel, int position)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].preemph.p->position = position;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+*/

wdsp/emph.h

@@ -0,0 +1,116 @@
+/*  emph.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2014, 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
+
+*/
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Partitioned Overlap-Save FM Pre-Emphasis                                *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _emphp_h
+#define _emphp_h
+#include "firmin.h"
+typedef struct _emphp
+{
+    int run;
+    int position;
+    int size;
+    int nc;
+    int mp;
+    double* in;
+    double* out;
+    int ctype;
+    double f_low;
+    double f_high;
+    double rate;
+    FIRCORE p;
+} emphp, *EMPHP;
+
+extern EMPHP create_emphp (int run, int position, int size, int nc, int mp,
+    double* in, double* out, int rate, int ctype, double f_low, double f_high);
+
+extern void destroy_emphp (EMPHP a);
+
+extern void flush_emphp (EMPHP a);
+
+extern void xemphp (EMPHP a, int position);
+
+extern void setBuffers_emphp (EMPHP a, double* in, double* out);
+
+extern void setSamplerate_emphp (EMPHP a, int rate);
+
+extern void setSize_emphp (EMPHP a, int size);
+
+__declspec (dllexport) void SetTXAFMEmphMP (int channel, int mp);
+
+__declspec (dllexport) void SetTXAFMEmphNC (int channel, int nc);
+
+__declspec (dllexport) void SetTXAFMPreEmphFreqs(int channel, double low, double high);
+
+#endif
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                       Overlap-Save FM Pre-Emphasis                                    *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _emph_h
+#define _emph_h
+
+typedef struct _emph
+{
+    int run;
+    int position;
+    int size;
+    double* in;
+    double* out;
+    int ctype;
+    double f_low;
+    double f_high;
+    double* infilt;
+    double* product;
+    double* mults;
+    double rate;
+    fftw_plan CFor;
+    fftw_plan CRev;
+} emph, *EMPH;
+
+extern EMPH create_emph (int run, int position, int size, double* in, double* out, int rate, int ctype, double f_low, double f_high);
+
+extern void destroy_emph (EMPH a);
+
+extern void flush_emph (EMPH a);
+
+extern void xemph (EMPH a, int position);
+
+extern void setBuffers_emph (EMPH a, double* in, double* out);
+
+extern void setSamplerate_emph (EMPH a, int rate);
+
+extern void setSize_emph (EMPH a, int size);
+
+#endif

wdsp/eq.c

@@ -0,0 +1,853 @@
+/*  eq.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2016, 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"
+
+int fEQcompare (const void * a, const void * b)
+{
+    if (*(double*)a < *(double*)b)
+        return -1;
+    else if (*(double*)a == *(double*)b)
+        return 0;
+    else
+        return 1;
+}
+
+double* eq_impulse (int N, int nfreqs, double* F, double* G, double samplerate, double scale, int ctfmode, int wintype)
+{
+    double* fp = (double *) malloc0 ((nfreqs + 2)   * sizeof (double));
+    double* gp = (double *) malloc0 ((nfreqs + 2)   * sizeof (double));
+    double* A  = (double *) malloc0 ((N / 2 + 1) * sizeof (double));
+    double* sary = (double *) malloc0 (2 * nfreqs * sizeof (double));
+    double gpreamp, f, frac;
+    double* impulse;
+    int i, j, mid;
+    fp[0] = 0.0;
+    fp[nfreqs + 1] = 1.0;
+    gpreamp = G[0];
+    for (i = 1; i <= nfreqs; i++)
+    {
+        fp[i] = 2.0 * F[i] / samplerate;
+        if (fp[i] < 0.0) fp[i] = 0.0;
+        if (fp[i] > 1.0) fp[i] = 1.0;
+        gp[i] = G[i];
+    }
+    for (i = 1, j = 0; i <= nfreqs; i++, j+=2)
+    {
+        sary[j + 0] = fp[i];
+        sary[j + 1] = gp[i];
+    }
+    qsort (sary, nfreqs, 2 * sizeof (double), fEQcompare);
+    for (i = 1, j = 0; i <= nfreqs; i++, j+=2)
+    {
+        fp[i] = sary[j + 0];
+        gp[i] = sary[j + 1];
+    }
+    gp[0] = gp[1];
+    gp[nfreqs + 1] = gp[nfreqs];
+    mid = N / 2;
+    j = 0;
+    if (N & 1)
+    {
+        for (i = 0; i <= mid; i++)
+        {
+            f = (double)i / (double)mid;
+            while (f > fp[j + 1]) j++;
+            frac = (f - fp[j]) / (fp[j + 1] - fp[j]);
+            A[i] = pow (10.0, 0.05 * (frac * gp[j + 1] + (1.0 - frac) * gp[j] + gpreamp)) * scale;
+        }
+    }
+    else
+    {
+        for (i = 0; i < mid; i++)
+        {
+            f = ((double)i + 0.5) / (double)mid;
+            while (f > fp[j + 1]) j++;
+            frac = (f - fp[j]) / (fp[j + 1] - fp[j]);
+            A[i] = pow (10.0, 0.05 * (frac * gp[j + 1] + (1.0 - frac) * gp[j] + gpreamp)) * scale;
+        }
+    }
+    if (ctfmode == 0)
+    {
+        int k, low, high;
+        double lowmag, highmag, flow4, fhigh4;
+        if (N & 1)
+        {
+            low = (int)(fp[1] * mid);
+            high = (int)(fp[nfreqs] * mid + 0.5);
+            lowmag = A[low];
+            highmag = A[high];
+            flow4 = pow((double)low / (double)mid, 4.0);
+            fhigh4 = pow((double)high / (double)mid, 4.0);
+            k = low;
+            while (--k >= 0)
+            {
+                f = (double)k / (double)mid;
+                lowmag *= (f * f * f * f) / flow4;
+                if (lowmag < 1.0e-100) lowmag = 1.0e-100;
+                A[k] = lowmag;
+            }
+            k = high;
+            while (++k <= mid)
+            {
+                f = (double)k / (double)mid;
+                highmag *= fhigh4 / (f * f * f * f);
+                if (highmag < 1.0e-100) highmag = 1.0e-100;
+                A[k] = highmag;
+            }
+        }
+        else
+        {
+            low = (int)(fp[1] * mid - 0.5);
+            high = (int)(fp[nfreqs] * mid - 0.5);
+            lowmag = A[low];
+            highmag = A[high];
+            flow4 = pow((double)low / (double)mid, 4.0);
+            fhigh4 = pow((double)high / (double)mid, 4.0);
+            k = low;
+            while (--k >= 0)
+            {
+                f = (double)k / (double)mid;
+                lowmag *= (f * f * f * f) / flow4;
+                if (lowmag < 1.0e-100) lowmag = 1.0e-100;
+                A[k] = lowmag;
+            }
+            k = high;
+            while (++k < mid)
+            {
+                f = (double)k / (double)mid;
+                highmag *= fhigh4 / (f * f * f * f);
+                if (highmag < 1.0e-100) highmag = 1.0e-100;
+                A[k] = highmag;
+            }
+        }
+    }
+    if (N & 1)
+        impulse = fir_fsamp_odd(N, A, 1, 1.0, wintype);
+    else
+        impulse = fir_fsamp(N, A, 1, 1.0, wintype);
+    // print_impulse("eq.txt", N, impulse, 1, 0);
+    _aligned_free (sary);
+    _aligned_free (A);
+    _aligned_free (gp);
+    _aligned_free (fp);
+    return impulse;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Equalizer                                  *
+*                                                                                                       *
+********************************************************************************************************/
+
+EQP create_eqp (int run, int size, int nc, int mp, double *in, double *out, int nfreqs, double* F, double* G, int ctfmode, int wintype, int samplerate)
+{
+    // NOTE:  'nc' must be >= 'size'
+    EQP a = (EQP) malloc0 (sizeof (eqp));
+    double* impulse;
+    a->run = run;
+    a->size = size;
+    a->nc = nc;
+    a->mp = mp;
+    a->in = in;
+    a->out = out;
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    a->ctfmode = ctfmode;
+    a->wintype = wintype;
+    a->samplerate = (double)samplerate;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    a->p = create_fircore (a->size, a->in, a->out, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+    return a;
+}
+
+void destroy_eqp (EQP a)
+{
+    destroy_fircore (a->p);
+    _aligned_free (a);
+}
+
+void flush_eqp (EQP a)
+{
+    flush_fircore (a->p);
+}
+
+void xeqp (EQP a)
+{
+    if (a->run)
+        xfircore (a->p);
+    else
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_eqp (EQP a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    setBuffers_fircore (a->p, a->in, a->out);
+}
+
+void setSamplerate_eqp (EQP a, int rate)
+{
+    double* impulse;
+    a->samplerate = rate;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+void setSize_eqp (EQP a, int size)
+{
+    double* impulse;
+    a->size = size;
+    setSize_fircore (a->p, a->size);
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                           Partitioned Overlap-Save Equalizer:  RXA Properties                         *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXAEQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].eqp.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAEQNC (int channel, int nc)
+{
+    EQP a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eqp.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAEQMP (int channel, int mp)
+{
+    EQP a;
+    a = rxa[channel].eqp.p;
+    if (a->mp != mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}
+
+PORT
+void SetRXAEQProfile (int channel, int nfreqs, double* F, double* G)
+{
+    EQP a;
+    double* impulse;
+    a = rxa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G,
+        a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetRXAEQCtfmode (int channel, int mode)
+{
+    EQP a;
+    double* impulse;
+    a = rxa[channel].eqp.p;
+    a->ctfmode = mode;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetRXAEQWintype (int channel, int wintype)
+{
+    EQP a;
+    double* impulse;
+    a = rxa[channel].eqp.p;
+    a->wintype = wintype;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetRXAGrphEQ (int channel, int *rxeq)
+{   // three band equalizer (legacy compatibility)
+    EQP a;
+    double* impulse;
+    a = rxa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 4;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1] =  150.0;
+    a->F[2] =  400.0;
+    a->F[3] = 1500.0;
+    a->F[4] = 6000.0;
+    a->G[0] = (double)rxeq[0];
+    a->G[1] = (double)rxeq[1];
+    a->G[2] = (double)rxeq[1];
+    a->G[3] = (double)rxeq[2];
+    a->G[4] = (double)rxeq[3];
+    a->ctfmode = 0;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetRXAGrphEQ10 (int channel, int *rxeq)
+{   // ten band equalizer (legacy compatibility)
+    EQP a;
+    double* impulse;
+    int i;
+    a = rxa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 10;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1]  =    32.0;
+    a->F[2]  =    63.0;
+    a->F[3]  =   125.0;
+    a->F[4]  =   250.0;
+    a->F[5]  =   500.0;
+    a->F[6]  =  1000.0;
+    a->F[7]  =  2000.0;
+    a->F[8]  =  4000.0;
+    a->F[9]  =  8000.0;
+    a->F[10] = 16000.0;
+    for (i = 0; i <= a->nfreqs; i++)
+        a->G[i] = (double)rxeq[i];
+    a->ctfmode = 0;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    // print_impulse ("rxeq.txt", a->nc, impulse, 1, 0);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                           Partitioned Overlap-Save Equalizer:  TXA Properties                         *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXAEQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].eqp.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAEQNC (int channel, int nc)
+{
+    EQP a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eqp.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAEQMP (int channel, int mp)
+{
+    EQP a;
+    a = txa[channel].eqp.p;
+    if (a->mp != mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}
+
+PORT
+void SetTXAEQProfile (int channel, int nfreqs, double* F, double* G)
+{
+    EQP a;
+    double* impulse;
+    a = txa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetTXAEQCtfmode (int channel, int mode)
+{
+    EQP a;
+    double* impulse;
+    a = txa[channel].eqp.p;
+    a->ctfmode = mode;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetTXAEQWintype (int channel, int wintype)
+{
+    EQP a;
+    double* impulse;
+    a = txa[channel].eqp.p;
+    a->wintype = wintype;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetTXAGrphEQ (int channel, int *txeq)
+{   // three band equalizer (legacy compatibility)
+    EQP a;
+    double* impulse;
+    a = txa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 4;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1] =  150.0;
+    a->F[2] =  400.0;
+    a->F[3] = 1500.0;
+    a->F[4] = 6000.0;
+    a->G[0] = (double)txeq[0];
+    a->G[1] = (double)txeq[1];
+    a->G[2] = (double)txeq[1];
+    a->G[3] = (double)txeq[2];
+    a->G[4] = (double)txeq[3];
+    a->ctfmode = 0;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+PORT
+void SetTXAGrphEQ10 (int channel, int *txeq)
+{   // ten band equalizer (legacy compatibility)
+    EQP a;
+    double* impulse;
+    int i;
+    a = txa[channel].eqp.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 10;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1]  =    32.0;
+    a->F[2]  =    63.0;
+    a->F[3]  =   125.0;
+    a->F[4]  =   250.0;
+    a->F[5]  =   500.0;
+    a->F[6]  =  1000.0;
+    a->F[7]  =  2000.0;
+    a->F[8]  =  4000.0;
+    a->F[9]  =  8000.0;
+    a->F[10] = 16000.0;
+    for (i = 0; i <= a->nfreqs; i++)
+        a->G[i] = (double)txeq[i];
+    a->ctfmode = 0;
+    impulse = eq_impulse (a->nc, a->nfreqs, a->F, a->G, a->samplerate, 1.0 / (2.0 * a->size), a->ctfmode, a->wintype);
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           Overlap-Save Equalizer                                      *
+*                                                                                                       *
+********************************************************************************************************/
+
+
+double* eq_mults (int size, int nfreqs, double* F, double* G, double samplerate, double scale, int ctfmode, int wintype)
+{
+    double* impulse = eq_impulse (size + 1, nfreqs, F, G, samplerate, scale, ctfmode, wintype);
+    double* mults = fftcv_mults(2 * size, impulse);
+    _aligned_free (impulse);
+    return mults;
+}
+
+void calc_eq (EQ a)
+{
+    a->scale = 1.0 / (double)(2 * a->size);
+    a->infilt = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->product = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->CFor = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->infilt, (fftw_complex *)a->product, FFTW_FORWARD, FFTW_PATIENT);
+    a->CRev = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->product, (fftw_complex *)a->out, FFTW_BACKWARD, FFTW_PATIENT);
+    a->mults = eq_mults(a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+}
+
+void decalc_eq (EQ a)
+{
+    fftw_destroy_plan(a->CRev);
+    fftw_destroy_plan(a->CFor);
+    _aligned_free(a->mults);
+    _aligned_free(a->product);
+    _aligned_free(a->infilt);
+}
+
+EQ create_eq (int run, int size, double *in, double *out, int nfreqs, double* F, double* G, int ctfmode, int wintype, int samplerate)
+{
+    EQ a = (EQ) malloc0 (sizeof (eq));
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    a->ctfmode = ctfmode;
+    a->wintype = wintype;
+    a->samplerate = (double)samplerate;
+    calc_eq (a);
+    return a;
+}
+
+void destroy_eq (EQ a)
+{
+    decalc_eq (a);
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    _aligned_free (a);
+}
+
+void flush_eq (EQ a)
+{
+    memset (a->infilt, 0, 2 * a->size * sizeof (complex));
+}
+
+void xeq (EQ a)
+{
+    int i;
+    double I, Q;
+    if (a->run)
+    {
+        memcpy (&(a->infilt[2 * a->size]), a->in, a->size * sizeof (complex));
+        fftw_execute (a->CFor);
+        for (i = 0; i < 2 * a->size; i++)
+        {
+            I = a->product[2 * i + 0];
+            Q = a->product[2 * i + 1];
+            a->product[2 * i + 0] = I * a->mults[2 * i + 0] - Q * a->mults[2 * i + 1];
+            a->product[2 * i + 1] = I * a->mults[2 * i + 1] + Q * a->mults[2 * i + 0];
+        }
+        fftw_execute (a->CRev);
+        memcpy (a->infilt, &(a->infilt[2 * a->size]), a->size * sizeof(complex));
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_eq (EQ a, double* in, double* out)
+{
+    decalc_eq (a);
+    a->in = in;
+    a->out = out;
+    calc_eq (a);
+}
+
+void setSamplerate_eq (EQ a, int rate)
+{
+    decalc_eq (a);
+    a->samplerate = rate;
+    calc_eq (a);
+}
+
+void setSize_eq (EQ a, int size)
+{
+    decalc_eq (a);
+    a->size = size;
+    calc_eq (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Overlap-Save Equalizer:  RXA Properties                                 *
+*                                                                                                       *
+********************************************************************************************************/
+/*  // UNCOMMENT properties when a pointer is in place in rxa[channel]
+PORT
+void SetRXAEQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].eq.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAEQProfile (int channel, int nfreqs, double* F, double* G)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAEQCtfmode (int channel, int mode)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eq.p;
+    a->ctfmode = mode;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAEQWintype (int channel, int wintype)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eq.p;
+    a->wintype = wintype;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAGrphEQ (int channel, int *rxeq)
+{   // three band equalizer (legacy compatibility)
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 4;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1] =  150.0;
+    a->F[2] =  400.0;
+    a->F[3] = 1500.0;
+    a->F[4] = 6000.0;
+    a->G[0] = (double)rxeq[0];
+    a->G[1] = (double)rxeq[1];
+    a->G[2] = (double)rxeq[1];
+    a->G[3] = (double)rxeq[2];
+    a->G[4] = (double)rxeq[3];
+    a->ctfmode = 0;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAGrphEQ10 (int channel, int *rxeq)
+{   // ten band equalizer (legacy compatibility)
+    EQ a;
+    int i;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 10;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1]  =    32.0;
+    a->F[2]  =    63.0;
+    a->F[3]  =   125.0;
+    a->F[4]  =   250.0;
+    a->F[5]  =   500.0;
+    a->F[6]  =  1000.0;
+    a->F[7]  =  2000.0;
+    a->F[8]  =  4000.0;
+    a->F[9]  =  8000.0;
+    a->F[10] = 16000.0;
+    for (i = 0; i <= a->nfreqs; i++)
+        a->G[i] = (double)rxeq[i];
+    a->ctfmode = 0;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+*/
+/********************************************************************************************************
+*                                                                                                       *
+*                               Overlap-Save Equalizer:  TXA Properties                                 *
+*                                                                                                       *
+********************************************************************************************************/
+/*  // UNCOMMENT properties when a pointer is in place in rxa[channel]
+PORT
+void SetTXAEQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].eq.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAEQProfile (int channel, int nfreqs, double* F, double* G)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = nfreqs;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    memcpy (a->F, F, (nfreqs + 1) * sizeof (double));
+    memcpy (a->G, G, (nfreqs + 1) * sizeof (double));
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAEQCtfmode (int channel, int mode)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eq.p;
+    a->ctfmode = mode;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAEQMethod (int channel, int wintype)
+{
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eq.p;
+    a->wintype = wintype;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAGrphEQ (int channel, int *txeq)
+{   // three band equalizer (legacy compatibility)
+    EQ a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 4;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1] =  150.0;
+    a->F[2] =  400.0;
+    a->F[3] = 1500.0;
+    a->F[4] = 6000.0;
+    a->G[0] = (double)txeq[0];
+    a->G[1] = (double)txeq[1];
+    a->G[2] = (double)txeq[1];
+    a->G[3] = (double)txeq[2];
+    a->G[4] = (double)txeq[3];
+    a->ctfmode = 0;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAGrphEQ10 (int channel, int *txeq)
+{   // ten band equalizer (legacy compatibility)
+    EQ a;
+    int i;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].eq.p;
+    _aligned_free (a->G);
+    _aligned_free (a->F);
+    a->nfreqs = 10;
+    a->F = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->G = (double *) malloc0 ((a->nfreqs + 1) * sizeof (double));
+    a->F[1]  =    32.0;
+    a->F[2]  =    63.0;
+    a->F[3]  =   125.0;
+    a->F[4]  =   250.0;
+    a->F[5]  =   500.0;
+    a->F[6]  =  1000.0;
+    a->F[7]  =  2000.0;
+    a->F[8]  =  4000.0;
+    a->F[9]  =  8000.0;
+    a->F[10] = 16000.0;
+    for (i = 0; i <= a->nfreqs; i++)
+        a->G[i] = (double)txeq[i];
+    a->ctfmode = 0;
+    _aligned_free (a->mults);
+    a->mults = eq_mults (a->size, a->nfreqs, a->F, a->G, a->samplerate, a->scale, a->ctfmode, a->wintype);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+*/

wdsp/eq.h

@@ -0,0 +1,127 @@
+/*  eq.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 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
+
+*/
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Equalizer                                  *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _eqp_h
+#define _eqp_h
+#include "firmin.h"
+typedef struct _eqp
+{
+    int run;
+    int size;
+    int nc;
+    int mp;
+    double* in;
+    double* out;
+    int nfreqs;
+    double* F;
+    double* G;
+    int ctfmode;
+    int wintype;
+    double samplerate;
+    FIRCORE p;
+} eqp, *EQP;
+
+extern double* eq_impulse (int N, int nfreqs, double* F, double* G, double samplerate, double scale, int ctfmode, int wintype);
+
+extern EQP create_eqp (int run, int size, int nc, int mp, double *in, double *out,
+    int nfreqs, double* F, double* G, int ctfmode, int wintype, int samplerate);
+
+extern void destroy_eqp (EQP a);
+
+extern void flush_eqp (EQP a);
+
+extern void xeqp (EQP a);
+
+extern void setBuffers_eqp (EQP a, double* in, double* out);
+
+extern void setSamplerate_eqp (EQP a, int rate);
+
+extern void setSize_eqp (EQP a, int size);
+
+__declspec (dllexport) void SetRXAEQNC (int channel, int nc);
+
+__declspec (dllexport) void SetRXAEQMP (int channel, int mp);
+
+__declspec (dllexport) void SetTXAEQNC (int channel, int nc);
+
+__declspec (dllexport) void SetTXAEQMP (int channel, int mp);
+
+#endif
+
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           Overlap-Save Equalizer                                      *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _eq_h
+#define _eq_h
+
+typedef struct _eq
+{
+    int run;
+    int size;
+    double* in;
+    double* out;
+    int nfreqs;
+    double* F;
+    double* G;
+    double* infilt;
+    double* product;
+    double* mults;
+    double scale;
+    int ctfmode;
+    int wintype;
+    double samplerate;
+    fftw_plan CFor;
+    fftw_plan CRev;
+}eq, *EQ;
+
+extern double* eq_mults (int size, int nfreqs, double* F, double* G, double samplerate, double scale, int ctfmode, int wintype);
+
+extern EQ create_eq (int run, int size, double *in, double *out, int nfreqs, double* F, double* G, int ctfmode, int wintype, int samplerate);
+
+extern void destroy_eq (EQ a);
+
+extern void flush_eq (EQ a);
+
+extern void xeq (EQ a);
+
+extern void setBuffers_eq (EQ a, double* in, double* out);
+
+extern void setSamplerate_eq (EQ a, int rate);
+
+extern void setSize_eq (EQ a, int size);
+
+#endif

wdsp/fcurve.c

@@ -0,0 +1,154 @@
+/*  fcurve.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 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
+
+*/
+
+#include "comm.h"
+
+double* fc_impulse (int nc, double f0, double f1, double g0, double g1, int curve, double samplerate, double scale, int ctfmode, int wintype)
+{
+    double* A  = (double *) malloc0 ((nc / 2 + 1) * sizeof (double));
+    int i;
+    double fn, f;
+    double* impulse;
+    int mid = nc / 2;
+    double g0_lin = pow(10.0, g0 / 20.0);
+    if (nc & 1)
+    {
+        for (i = 0; i <= mid; i++)
+        {
+            fn = (double)i / (double)mid;
+            f = fn * samplerate / 2.0;
+            switch (curve)
+            {
+            case 0: // fm pre-emphasis
+                if (f0 > 0.0)
+                    A[i] = scale * (g0_lin * f / f0);
+                else
+                    A[i] = 0.0;
+                break;
+            case 1: // fm de-emphasis
+                if (f > 0.0)
+                    A[i] = scale * (g0_lin * f0 / f);
+                else
+                    A[i] = 0.0;
+                break;
+            }
+        }
+    }
+    else
+    {
+        for (i = 0; i < mid; i++)
+        {
+            fn = ((double)i + 0.5) / (double)mid;
+            f = fn * samplerate / 2.0;
+            switch (curve)
+            {
+            case 0: // fm pre-emphasis
+                if (f0 > 0.0)
+                    A[i] = scale * (g0_lin * f / f0);
+                else
+                    A[i] = 0.0;
+                break;
+            case 1: // fm de-emphasis
+                if (f > 0.0)
+                    A[i] = scale * (g0_lin * f0 / f);
+                else
+                    A[i] = 0.0;
+                break;
+            }
+        }
+    }
+    if (ctfmode == 0)
+    {
+        int k, low, high;
+        double lowmag, highmag, flow4, fhigh4;
+        if (nc & 1)
+        {
+            low  = (int)(2.0 * f0 / samplerate * mid);
+            high = (int)(2.0 * f1 / samplerate * mid + 0.5);
+            lowmag = A[low];
+            highmag = A[high];
+            flow4 = pow((double)low / (double)mid, 4.0);
+            fhigh4 = pow((double)high / (double)mid, 4.0);
+            k = low;
+            while (--k >= 0)
+            {
+                f = (double)k / (double)mid;
+                lowmag *= (f * f * f * f) / flow4;
+                if (lowmag < 1.0e-100) lowmag = 1.0e-100;
+                A[k] = lowmag;
+            }
+            k = high;
+            while (++k <= mid)
+            {
+                f = (double)k / (double)mid;
+                highmag *= fhigh4 / (f * f * f * f);
+                if (highmag < 1.0e-100) highmag = 1.0e-100;
+                A[k] = highmag;
+            }
+        }
+        else
+        {
+            low  = (int)(2.0 * f0 / samplerate * mid - 0.5);
+            high = (int)(2.0 * f1 / samplerate * mid - 0.5);
+            lowmag = A[low];
+            highmag = A[high];
+            flow4 = pow((double)low / (double)mid, 4.0);
+            fhigh4 = pow((double)high / (double)mid, 4.0);
+            k = low;
+            while (--k >= 0)
+            {
+                f = (double)k / (double)mid;
+                lowmag *= (f * f * f * f) / flow4;
+                if (lowmag < 1.0e-100) lowmag = 1.0e-100;
+                A[k] = lowmag;
+            }
+            k = high;
+            while (++k < mid)
+            {
+                f = (double)k / (double)mid;
+                highmag *= fhigh4 / (f * f * f * f);
+                if (highmag < 1.0e-100) highmag = 1.0e-100;
+                A[k] = highmag;
+            }
+        }
+    }
+    if (nc & 1)
+        impulse = fir_fsamp_odd(nc, A, 1, 1.0, wintype);
+    else
+        impulse = fir_fsamp(nc, A, 1, 1.0, wintype);
+    // print_impulse ("emph.txt", size + 1, impulse, 1, 0);
+    _aligned_free (A);
+    return impulse;
+}
+
+// generate mask for Overlap-Save Filter
+double* fc_mults (int size, double f0, double f1, double g0, double g1, int curve, double samplerate, double scale, int ctfmode, int wintype)
+{
+    double* impulse = fc_impulse (size + 1, f0, f1, g0, g1, curve, samplerate, scale, ctfmode, wintype);
+    double* mults = fftcv_mults(2 * size, impulse);
+    _aligned_free (impulse);
+    return mults;
+}