Quisk version 4.2.50

CHANGELOG.txt

@@ -1,3 +1,85 @@
+Quisk Version 4.2.50  December 2025
+===================================
+This is a quick fix for a bug in 4.2.49 that results in the repeated message
+AttributeError: 'Hardware' object has no attribute 'is_HermesLite2'
+
+Quisk Version 4.2.49  December 2025
+===================================
+This version adds more features for the Anan 10E and other OpenHPSDR radios. Please test.
+
+I added the forward power, SWR, temperature and PA current display to the OpenHPSDR radio. But
+I don't have that radio so I need some help. I think the forward and reverse power connections
+are correct, but it is still necessary to create a table to convert analog input codes to power.
+Go to the Hardware screen and check the "Power meter calibration" setting. You will need to enter
+"New" and create a suitable table. To see the ADC values, choose table "Show ADC".
+
+There is no protocol standard to specify which analog input is used for temperature and PA current.
+And AFAIK the Anan 10E has no such sensors. I am hoping to use the board ID to find the right input.
+The ADC conversion is linear and the two points are specified on the Hardware screen.
+
+I currently assigned the analog inputs as follows:
+AIN1	Forward power
+AIN2	Reverse power
+AIN3	PA current
+AIN5	Temperature
+If you need to see the ADC codes 0-4095 for each input, go to hermes/quisk_hardware.py and
+change DEBUG_SENSORS to one.
+
+Quisk Version 4.2.48  November 2025
+===================================
+This is a quisk fix for a bug in the new OpenHPSDR radios. It fixes use_rx_udp that was reset to zero.
+
+Quisk Version 4.2.47  November 2025
+===================================
+To improve the performance of Quisk with the Anan 10-E and other OpenHPSDR radios, I added
+a new type of radio "OpenHPSDR". Use this type for the Anan 10-E and similar radios. It removes
+all the protocol extensions added by the Hermes-Lite2 project.  I removed the configuration
+option "Hermes Lite2 protocol enable". This is now automatic.
+
+I added buttons for antennas 0, 1, 2 and 3 for OpenHPSDR radios that support more antennas.
+I added controls for the OpenHPSDR preamplifier and attenuator.
+
+I do not have any Apache Labs radios, so I must depend on my users to report problems and solutions.
+
+Quisk Version 4.2.46  October 2025
+==================================
+I fixed a bug in Hamlib rig 2 which happens with the new Pure Signal code. The Hamlib "f" command
+now always returns the main receiver frequency, not one of the added receiver frequencies. This is
+consistent with the Flex ZZ commands.
+
+Quisk Version 4.2.45  October 2025
+==================================
+The new FreeDV including RADE mode is now a built-in mode of Quisk. Just press "FDV" to talk FreeDV.
+In brief, install FreeDV-GUI from the web site. Then configure the usual Quisk digital input and
+output that you use for digital modes. In FreeDV-GUI use the digital input and output. For the
+speakers and microphone use the same ones as Quisk. Don't forget to configure your call sign in
+FreeDV so it works with FreeDV Reporter. For detailed instructions see http://james.ahlstrom.name/rade.html.
+
+The Quisk digital output device is now always active; a DGT mode is not required.
+
+Quisk Version 4.2.44  September 2025
+====================================
+Quisk now has Pure Signal for the Hermes Lite2. This is baseband predistortion to reduce IMD products.
+To use it, press "PsCal" for calibration, and "PS" to enable Pure Signal. There are no adjustments needed.
+More details are in the documentation.
+
+Quisk Version 4.2.43  September 2025
+====================================
+I fixed a bug in the WDSP library NR2 noise reduction feature. I added the "Trained" method of NR2.
+Quisk uses the WDSP library for the NR2 noise reduction and the SNB noise blanker. The WDSP library must be available
+for these features to work. For windows, the library is included. For Linux you must build it by changing to the
+quisk/wdsp directory and entering "make". The library is used in-place and no installation is necessary.
+The new "trained" method of NR2 is included. Press the NR2 button twice to select "NR2t".
+
+I added documentation for the new FreeDV mode RADE. See http://james.ahlstrom.name/rade.html.
+The default IMD magnitude is now 100%.
+
+Quisk Version 4.2.42  April 2025
+================================
+I fixed a bug in the Kenwood IF CAT command. Thanks Nick!  I moved the Pulse server IP address to the Sound screen.
+I added a CAT method to imitate the Elecraft K4 command set over a TCP port. Quisk now has four CAT controls. To
+make this clearer I added more documentation to the Remote screen.
+
 Quisk Version 4.2.41  January 2025
 ==================================
 I made some improvements to the top-level Config/Radios screen. There is now an option to

__init__.py

@@ -1,2 +1,2 @@
-#Quisk version 4.2.41
+#Quisk version 4.2.50
 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)

configure.py

@@ -88,6 +88,7 @@ class Configuration:
       platform_ignore = 'win_'
     self.sections = []
     self.receiver_data = []
+    self.receiver_explain = []
     self.StatePath = conf.settings_file_path
     if not self.StatePath:
       self.StatePath = os.path.join(app.QuiskFilesDir, "quisk_settings.json")
@@ -158,9 +159,9 @@ class Configuration:
         dlg.Destroy()
     else:
       radio_dict["use_soapy"] = '0'
-    if radio_type not in ("HiQSDR", "Hermes", "Red Pitaya", "Odyssey", "Odyssey2"):
+    if radio_type not in ("HiQSDR", "Hermes", "Red Pitaya", "Odyssey", "Odyssey2", "OpenHPSDR"):
       radio_dict["use_rx_udp"] = '0'
-    if radio_type in ("Hermes", "Red Pitaya", "Odyssey2"):
+    if radio_type in ("Hermes", "Red Pitaya", "Odyssey2", "OpenHPSDR"):
       if "Hermes_BandDict" not in radio_dict:
         radio_dict["Hermes_BandDict"] = {}
       if "Hermes_BandDictTx" not in radio_dict:
@@ -601,6 +602,7 @@ class Configuration:
     # Receiver sections start with 16 #, "Receivers ", receiver name, explain
     # self.receiver_data is a list of [receiver_name, receiver_data]
     # receiver_data is a list of [data_name, text, fmt, help_text, values]
+    # self.receiver_explain is a list of the text in Receivers: the name, a comma and the explanation.
 
     # Variable names start with ## variable_name   variable_text, format
     #     The format is integer, number, text, boolean, integer choice, text choice, rfile
@@ -638,6 +640,7 @@ class Configuration:
         continue
       if line[0:27] == '################ Receivers ':
         section = 'Receivers'
+        self.receiver_explain.append(line[27:].strip())
         args = line[27:].split(',', 1)
         rxname = args[0].strip()
         section_data = []
@@ -1338,7 +1341,16 @@ class RadioNotebook(wx.Notebook):	# The second-level notebook for each radio nam
     self.AddPage(page, "Sound")
     self.pages.append(page)
     for section, names in local_conf.sections:
-      if section in ('Sound', 'Bands', 'Filters'):		# There is a special page for these sections
+      if section == 'Sound':		# There is a special page for these sections
+        continue
+      if section == 'Bands':
+        continue
+      if section == 'Filters':
+        continue
+      if section == 'Remote':
+        page = RadioRemote(self, radio_name, section, names)
+        self.AddPage(page, section)
+        self.pages.append(page)
         continue
       page = RadioSection(self, radio_name, section, names)
       self.AddPage(page, section)
@@ -1979,6 +1991,16 @@ class ControlMixin:
     btn.SetSizeHints(h, h, h, h)
     self.gbs.Add(btn, (self.row, col), flag=wx.ALIGN_CENTER_VERTICAL|wx.RIGHT, border=self.charx*border)
     return sld, btn
+  def AddTextHeading(self, key, text, help_text):
+    if key in text:		# Start of heading
+      self.NextRow()
+      self.SetFont(self.heading_font)
+      self.AddTextL(1, help_text, span=7)
+      self.SetFont(self.font)
+      self.NextRow()
+      self.col = 2
+      return key
+    return None
   def AddPopupMenuHelp(self, Qclass, col, text, btn_text, help_text, span1=1, span2=1, border=1):
     if text:
       txt = wx.StaticText(self, -1, text)
@@ -2279,8 +2301,8 @@ class ControlMixin:
       if self.radio_name == Settings[1]:	# changed for current radio
         if name in ('hot_key_ptt_toggle', 'hot_key_ptt_if_hidden', 'keyupDelay', 'cwTone', 'pulse_audio_verbose_output',
                     'start_cw_delay', 'start_ssb_delay', 'maximum_tx_secs', 'quisk_serial_cts', 'quisk_serial_dsr',
-                    'hot_key_ptt1', 'hot_key_ptt2', 'midi_ptt_toggle', 'TxRxSilenceMsec', 'hermes_lite2_enable',
-                    'fixed_tune_offset'):
+                    'hot_key_ptt1', 'hot_key_ptt2', 'midi_ptt_toggle', 'TxRxSilenceMsec',
+                    'fixed_tune_offset', 'calibrate_temp_20', 'calibrate_temp_40', 'calibrate_current_0', 'calibrate_current_1'):
           setattr(conf, name, x)
           application.ImmediateChange(name)
         elif name[0:4] in ('lin_', 'win_'):
@@ -2381,6 +2403,7 @@ class BaseWindow(wx.ScrolledWindow, ControlMixin):
   def __init__(self, parent):
     wx.ScrolledWindow.__init__(self, parent)
     ControlMixin.__init__(self, parent)
+    self.SetScrollRate(8, 8)
 
 class ConfigConfig(BaseWindow):
   def __init__(self, parent, width):
@@ -2491,7 +2514,6 @@ The CW message will then repeat. A time of zero means no repeat.'
     sl, btn = self.AddTextSliderHelp(4, "    Repeat secs %.1f  ", 0, 0, 100, self.OnPlayFileRepeat, help_text, span=2, scale=0.1)
     self.NextRow()
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def MakeFileButton(self, text, path, index, name, help_text):
     if index < 10:	# record buttons
       cb = self.AddCheckBox(4, text, self.OnCheckRecPlay)
@@ -2669,7 +2691,6 @@ class ConfigTxAudio(BaseWindow):
     application.CtrlTxAudioPreemph = sld
     self.NextRow()
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def OnTimer(self, event):
     if not self.tmp_playing:
       self.timer.Stop()
@@ -2718,8 +2739,8 @@ class Radios(BaseWindow):	# The "Radios" first-level page
     self.NextRow()
     rb = self.AddRadioButton(1, "Add a new radio with the general type", self.NewTop)
     choices = []
-    for name, data in local_conf.receiver_data:
-      choices.append(name)
+    for explain in local_conf.receiver_explain:
+      choices.append(explain)
     choices.sort()
     self.add_type = self.AddComboCtrl(2, '', choices=choices, no_edit=True)
     self.add_type.SetSelection(0)
@@ -2747,7 +2768,6 @@ class Radios(BaseWindow):	# The "Radios" first-level page
     self.apply_btn = self.AddPushButton(3, "Apply", self.OnBtnApply)
     self.NewRadioNames()
     self.NewTop()
-    self.SetScrollRate(1, 1)
   def NewTop(self, event=None):
     if event:
       btn = event.GetEventObject()
@@ -2791,7 +2811,7 @@ class Radios(BaseWindow):	# The "Radios" first-level page
     if not name or self.DuplicateName(name):
       return
     self.add_name.SetValue('')
-    typ = self.add_type.GetValue().strip()
+    typ = self.add_type.GetValue().split(',')[0].strip()
     if local_conf.AddRadio(name, typ):
       self.NewRadioNames()
       local_conf.settings_changed = True
@@ -2890,16 +2910,7 @@ class RadioSection(BaseWindow):		# The pages for each section in the second-leve
     radio_dict = local_conf.GetRadioDict(self.radio_name)
     radio_type = radio_dict['hardware_file_type']
     for name, text, fmt, help_text, values in self.names:
-      if name == 'remote_radio_password':
-        self.AddTextButtonHelp(col, text, "Change", self.OnChangePassword, help_text)
-        if col == 1:
-          col = 4
-        else:
-          col = 1
-          self.NextRow()
-      elif name == 'remote_radio_ip' and radio_type != "Control Head":
-        continue
-      elif name == 'favorites_file_path':
+      if name == 'favorites_file_path':
         self.favorites_path = radio_dict.get('favorites_file_path', '')
         row = self.row
         self.row = 1
@@ -2993,7 +3004,6 @@ class RadioSection(BaseWindow):		# The pages for each section in the second-leve
       self.AddColSpacer(2, 20)
       self.AddColSpacer(5, 20)
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def OnButtonChangeFavorites(self, event):
     if isinstance(event, (ChoiceCombo, ComboCtrl)):
       path = event.GetValue()
@@ -3087,14 +3097,6 @@ class RadioSection(BaseWindow):		# The pages for each section in the second-leve
       index = self.list_ctrl.GetFirstSelected()
       if index >= 0:
         self.list_ctrl.Select(index, 0)
-  def OnChangePassword(self, event):
-    text = local_conf.globals.get('remote_radio_password', '')
-    dlg = wx.TextEntryDialog(self, "Please enter a password for remote access",
-          "Password Entry", text)
-    if dlg.ShowModal() == wx.ID_OK:
-      local_conf.globals["remote_radio_password"] = dlg.GetValue()
-      local_conf.settings_changed = True
-      local_conf.SaveState()
 
 class RadioHardwareBase(BaseWindow):		# The Hardware page in the second-level notebook for each radio
   def __init__(self, parent, radio_name):
@@ -3126,7 +3128,6 @@ class RadioHardwareBase(BaseWindow):		# The Hardware page in the second-level no
     self.NextRow(7)
     self.AddColSpacer(2, 20)
     self.AddColSpacer(5, 20)
-    self.SetScrollRate(1, 1)
   def OnButtonChangeHardware(self, event):
     if isinstance(event, (ChoiceCombo, ComboCtrl)):
       path = event.GetValue()
@@ -3173,25 +3174,24 @@ class RadioHardware(RadioHardwareBase):		# The Hardware page in the second-level
     data_names = local_conf.GetReceiverData(radio_type)
     col = 1
     border = 2
-    hermes_board_id = 0
-    if radio_type == "Hermes":
-      try:
-        hermes_board_id = application.Hardware.hermes_board_id
-      except:
-        pass
+    try:
+      is_HermesLite2 = application.Hardware.is_HermesLite2
+    except:
+      is_HermesLite2 = False
     if radio_name == Settings[1] and hasattr(application.Hardware, "ProgramGateware"):
       help_text = "Choose an RBF file and program the Gateware (FPGA software) over Ethernet."
       self.AddTextButtonHelp(1, "Gateware Update", "Program from RBF file..", application.Hardware.ProgramGateware, help_text)
       col = 1
       self.NextRow(self.row + 2)
     for name, text, fmt, help_text, values in data_names:
+      text = text.replace("\\u2103", "\u2103")
       if name in ('hardware_file_name', 'widgets_file_name'):
         pass
       elif name[0:4] == platform_ignore:
         pass
       elif name in ('Hermes_BandDictEnTx', ):
         pass
-      elif 'Hl2_' in name and hermes_board_id != 6:
+      elif 'Hl2_' in name and not is_HermesLite2:
         pass
       elif fmt[0:4] in ('dict', 'list'):
         pass
@@ -3220,7 +3220,7 @@ class RadioHardware(RadioHardwareBase):		# The Hardware page in the second-level
           col = 1
           border = 2
           self.NextRow()
-    if hermes_board_id == 6:
+    if is_HermesLite2:
       if col == 4:
         self.NextRow()
       help_text = ('This controls the bias level for transistors in the final power amplifier.  Enter a level from 0 to 255.'
@@ -3236,7 +3236,6 @@ class RadioHardware(RadioHardwareBase):		# The Hardware page in the second-level
       self.HermesWriteBiasButton = self.AddPushButton(7, "Write", self.OnButtonHermesWriteBias, border=0)
       self.HermesWriteBiasButton.Enable(enbl)
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def OnHermesChangeBias0(self, event):
     value = self.HermesBias0.GetValue()
     application.Hardware.ChangeBias0(value)
@@ -3747,7 +3746,7 @@ class RadioSound(BaseWindow):		# The Sound page in the second-level notebook for
     self.radio_dict = local_conf.GetRadioDict(self.radio_name)
     self.num_cols = 8
     for name, text, fmt, help_text, values in local_conf.GetSectionData('Sound'):
-      if name in ('digital_output_level', 'file_play_level'):
+      if name in ('digital_output_level', 'file_play_level', 'IQ_Server_IP'):
         value = self.GetValue(name, self.radio_dict)
         no_edit = "choice" in fmt or fmt == 'boolean'
         txt, cb, btn = self.AddTextComboHelp(1, text, value, values, help_text, no_edit)
@@ -3850,7 +3849,6 @@ class RadioSound(BaseWindow):		# The Sound page in the second-level notebook for
         break
     self.gbs = gbs
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def ItemValue(self, row, col):
     data_name = self.sound_names[row][col]
     if col == 4:		# Device names
@@ -4078,7 +4076,6 @@ The Tx bits are sent as the "Alex" filters and are sent to the J16 interface.'
         self.AddBitField(col + 1, 7, 'Hermes_BandDictTx', band, bus, self.ChangeIO)
         self.NextRow()
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def SortCmp(self, item1):
     # Numerical conversion of band name to  megahertz
     try:
@@ -4182,6 +4179,87 @@ The Tx bits are sent as the "Alex" filters and are sent to the J16 interface.'
     if hasattr(application.Hardware, "ImmediateChange"):
       application.Hardware.ImmediateChange(name)
 
+class RadioRemote(BaseWindow):		# The Remote page in the second-level notebook for each radio
+  def __init__(self, parent, radio_name, section, names):
+    BaseWindow.__init__(self, parent)
+    self.radio_name = radio_name
+    self.section = section
+    self.names = names
+    self.heading_font = wx.Font(conf.config_font_size, wx.FONTFAMILY_SWISS, wx.ITALIC,
+          wx.FONTWEIGHT_NORMAL, False, conf.quisk_typeface)
+    self.MakeControls()
+  def MakeControls(self):
+    self.num_cols = 8
+    #self.MarkCols()
+    self.AddColSpacer(1, 3)
+    self.NextRow()
+    self.col = 2
+    radio_dict = local_conf.GetRadioDict(self.radio_name)
+    radio_type = radio_dict['hardware_file_type']
+    heading = None
+    for name, text, fmt, help_text, values in self.names:
+      if heading:
+        if heading not in text:		# End of heading
+          heading = None
+          self.NextRow()
+          self.col = 2
+      if not heading:
+        heading = self.AddTextHeading("Hamlib Rig 2", text,
+'Rig control commands using "Hamlib Rig 2" are sent to a Quisk TCP port from multiple clients.')
+      if not heading:
+        heading = self.AddTextHeading("XML-RPC", text,
+'XML-RPC is a protocol that allows clients to call methods on a remote server using XML passed via HTTP')
+      if not heading:
+        heading = self.AddTextHeading("CAT serial", text,
+'"CAT serial" supports the FlexRadio PowerSDR 2.x command set sent over a serial port. Some Kenwood commands are included.')
+      if not heading:
+        heading = self.AddTextHeading("remote radio", text,
+'Quisk can be used as a control head to control a real radio located remotely. This feature is due to AC2YD.')
+      if not heading:
+        heading = self.AddTextHeading("K4 TCP", text,
+'Rig control commands using "K4 TCP" use the Elecraft K4 command set, and are sent to a Quisk TCP port from multiple clients.')
+      if not heading:
+        heading = self.AddTextHeading("Dx cluster", text,
+'"Dx Cluster" is a telnet service used to announce Dx spots in real time.')
+      if name == 'remote_radio_password':
+        self.AddTextButtonHelp(self.col, text, "Change", self.OnChangePassword, help_text)
+        if self.col == 2:
+          self.col = 5
+        else:
+          self.col = 2
+          self.NextRow()
+      elif name == 'remote_radio_ip' and radio_type != "Control Head":
+        continue
+      else:
+        if fmt[0:4] in ('dict', 'list'):
+          continue
+        if name[0:4] == platform_ignore:
+          continue
+        value = self.GetValue(name, radio_dict)
+        no_edit = "choice" in fmt or fmt == 'boolean'
+        txt, cb, btn = self.AddTextComboHelp(self.col, text, value, values, help_text, no_edit)
+        cb.handler = self.OnChange
+        cb.quisk_data_name = name
+        if self.col == 2:
+          self.col = 5
+        else:
+          self.col = 2
+          self.NextRow()
+    self.NextRow()
+    self.AddTextL(1, "End of list", span=7)
+    self.AddColSpacer(3, 20)
+    self.AddColSpacer(6, 20)
+    self.FitInside()
+  def OnChangePassword(self, event):
+    text = local_conf.globals.get('remote_radio_password', '')
+    dlg = wx.TextEntryDialog(self, "Please enter a password for remote access",
+          "Password Entry", text)
+    if dlg.ShowModal() == wx.ID_OK:
+      local_conf.globals["remote_radio_password"] = dlg.GetValue()
+      local_conf.settings_changed = True
+      local_conf.SaveState()
+
+
 class xxRadioFilters(BaseWindow):		# The Filters page in the second-level notebook for each radio
   def __init__(self, parent, radio_name):
     BaseWindow.__init__(self, parent)
@@ -4253,7 +4331,6 @@ class xxRadioFilters(BaseWindow):		# The Filters page in the second-level notebo
       self.NextRow()
       index += 1
     self.FitInside()
-    self.SetScrollRate(1, 1)
   def OnChangeFreq(self, event):
     freq = event.GetString()
     radio_dict = local_conf.GetRadioDict(self.radio_name)

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>
@@ -197,7 +200,8 @@ If you have Python 3 installed on your computer, just continue to use that versi
 <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>
@@ -373,12 +377,13 @@ python3 quisk.py
 <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:
+You only clone github for the initial installation. To update Quisk, use pull followed by make:
 </p>
 <br>
 <pre>
 cd ~/quisk
 git pull
+make
 </pre>
 <br>
 <p>
@@ -1603,6 +1608,46 @@ 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

freedv.c

@@ -1,681 +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 {
-  int interleave_frames;  // now unused but remains to prevent breaking API for
-                          // legacy apps
-
-  // parameters for FREEDV_MODE_FSK_LDPC
-  int M;             // 2 or 4 FSK
-  int Rs;            // Symbol rate Hz
-  int Fs;            // Sample rate Hz
-  int first_tone;    // Freq of first tone Hz
-  int tone_spacing;  // Spacing between tones Hz
-  char *codename;    // LDPC codename, from codes listed in ldpc_codes.c
-};
-
-#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;
-int n_max_modem_samples = 1000;
-
-#define SPEECH_BUF_SIZE		12000		// speech buffer size increased from 3000
-#define MAX_MODEM_SAMPLES	8192		// maximum of n_max_modem_samples for any mode
-#define MAX_SPEECH_SAMPLES	8192		// maximum of n_speech_samples for any mode
-static struct _rx_channel{
-	struct freedv * hFreedv;
-	short * 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 = 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(short * sSamples, double * dsamples, int count, int bank)	// Called from the sound thread.
-{	// Input digital modulation is sSamples; 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;
-	struct freedv * hF;
-	struct _rx_channel * pCh;
-	int nCountOut = 1;	// In case we have a higher (or lower) value for o/p count
-
-	if (sSamples == 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++) {
-		pCh->demod_in[pCh->rxdata_index] = sSamples[i];
-		pCh->rxdata_index++;
-		if (pCh->rxdata_index >= need) {
-			if (pCh->speech_available + n_speech_samples < SPEECH_BUF_SIZE) {		// check for buffer space
-				have = freedv_rx(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] * ((double)CLIP32 / CLIP16 * 0.99);
-	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, int is_real)	// 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
-		return 0;	// nothing to do
-	}
-	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) * MAX_MODEM_SAMPLES);
-		memset(mod_out, 0, sizeof(COMP) * MAX_MODEM_SAMPLES);
-		speech_in = (short*)malloc(sizeof(short) * MAX_SPEECH_SAMPLES);
-		speech_index=0;
-		mod_index=0;
-		real_mod_out = (short *)malloc(sizeof(short) * MAX_MODEM_SAMPLES);
-		memset(real_mod_out, 0, sizeof(short) * MAX_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( is_real ) {
-				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( is_real ) {
-						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;
-		}
-	}
-	quisk_freedv_rx(NULL, NULL, 0, 0);
-	quisk_freedv_tx(NULL, NULL, 0, 0);
-	freedv_current_mode = -1;
-}
-
-static int OpenFreedv(void)	// Called from the GUI thread or sound thread
-{
-	int i;
-	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_64 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_700D || requested_mode == FREEDV_MODE_700E) && freedv_open_advanced) {
-       	struct freedv_advanced adv;
-	memset(&adv, 0, sizeof(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 = (short *)malloc(sizeof(short) * 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

@@ -1,24 +1 @@
-extern int freedv_current_mode;         // the current FreeDV running mode
-
-// available speech modes
-#define FREEDV_MODE_1600 0
-#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_700E 13
-#define FREEDV_MODE_2020 8
-#define FREEDV_MODE_2020B 16
-
-// available data modes
-#define FREEDV_MODE_FSK_LDPC 9
-#define FREEDV_MODE_DATAC1 10
-#define FREEDV_MODE_DATAC3 12
-#define FREEDV_MODE_DATAC0 14
-#define FREEDV_MODE_DATAC4 18
-#define FREEDV_MODE_DATAC13 19
-
-// Sample rates used
-#define FREEDV_FS_8000 8000
-#define FREEDV_FS_16000 16000
+// 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

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;
@@ -615,7 +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
@@ -624,140 +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.
-	// Modified by N2ADR January, 2024.
-	int i;
-	// The input sample rate is mic_sample_rate, either 8000 or 48000 sps.
-	// The FreeDV codec requires an input speech_sample_rate of either 8000 or 16000 sps depending on the mode. Input is always real.
-	// The FreeDV codec will output samples at modem_sample_rate of 8000 or 48000 sps depending on the mode, and output may be real or complex.
-	double dtmp, magn, dsample;
-	int modem_sample_rate, speech_sample_rate, use_filter, is_real;
-	static int samples_size = 0;
-	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 filter1, filter2;
-	static struct quisk_dFilter filtDecim48to16, filtDecim16to8;;
-	static struct quisk_cFilter cfiltInterp8to48;
-
-	if (do_init) {		// initialization
-		//QuiskWavWriteOpen(&hWav, "jim.wav", 3, 1, 4, 8000, 1.0 / CLIP16);
-		do_init = 0;
-		quisk_filt_cInit(&filter1, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));  // at 48000 sps pass 3000 stop 4000 times two plus 3000
-		quisk_filt_cInit(&filter2, quiskFilt53D2Coefs, sizeof(quiskFilt53D2Coefs)/sizeof(double));  // at 8000 sps pass 1500 stop 1800 times two plus 1500
-		quisk_filt_cInit(&cfiltInterp8to48, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
-		quisk_filt_dInit(&filtDecim48to16, quiskAudio24p3Coefs, sizeof(quiskAudio24p3Coefs)/sizeof(double));	// pass 6000 stop 8000
-		quisk_filt_dInit(&filtDecim16to8, quiskFilt16dec8Coefs, sizeof(quiskFilt16dec8Coefs)/sizeof(double));	// pass 3000 stop 4000
-		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;
-	}
-	switch (freedv_current_mode) {		// n_modem_sample_rate and n_speech_sample_rate may not be valid
-		case FREEDV_MODE_2400A:
-		case FREEDV_MODE_2400B:
-			use_filter = 1;		// center frequency 3000
-			is_real = 1;
-			modem_sample_rate = 48000;
-			speech_sample_rate = 8000;
-			break;
-		case FREEDV_MODE_2020:
-			use_filter = 0;
-			is_real = 0;
-			modem_sample_rate = 8000;
-			speech_sample_rate = 16000;
-			break;
-		case FREEDV_MODE_800XA:
-			use_filter = 2;		// center frequency 1500
-			is_real = 1;
-			modem_sample_rate = 8000;
-			speech_sample_rate = 8000;
-			break;
-		default:
-			use_filter = 0;
-			is_real = 0;
-			modem_sample_rate = 8000;
-			speech_sample_rate = 8000;
-			break;
-	}
-	if ( ! filtered) {	// rxMode changed
-		quisk_filt_tune((struct quisk_dFilter *)&filter1, 3000.0 / 48000.0, rxMode != FDV_L);
-		quisk_filt_tune((struct quisk_dFilter *)&filter2, 1500.0 / 8000.0, rxMode != FDV_L);
-		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) {
-		switch (speech_sample_rate) {
-		case 16000:
-			count = quisk_dDecimate(dsamples, count, &filtDecim48to16, 3);
-			break;
-		case 8000:
-			count = quisk_dDecimate(dsamples, count, &filtDecim48to16, 3);
-			count = quisk_dDecimate(dsamples, count, &filtDecim16to8, 2);
-			break;
-		default:
-			QuiskPrintf("Failure to convert speech input rate in tx_filter_freedv\n");
-			break;
-		}
-	}
-	else if (quisk_sound_state.mic_sample_rate == 8000 && 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 / 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, is_real);
-	//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
-	if (use_filter == 1)	// convert float samples to complex with an analytic filter
-		count = quisk_cCDecimate(filtered, count, &filter1, 1);
-	else if (use_filter == 2)
-		count = quisk_cCDecimate(filtered, count, &filter2, 1);
-	// Interpolate up to 48000
-	if (MIC_OUT_RATE != modem_sample_rate ) 
-		count = quisk_cInterpolate(filtered, count, &cfiltInterp8to48, MIC_OUT_RATE / 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
@@ -894,6 +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]);
@@ -909,6 +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;
@@ -916,6 +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) {
@@ -928,6 +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);
@@ -971,15 +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;
@@ -1028,15 +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;
 	}
@@ -1102,12 +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;
@@ -1319,6 +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;
@@ -1326,11 +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;
 		}
@@ -1421,11 +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;
@@ -1447,14 +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
@@ -1578,7 +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);
 }
 
@@ -1624,3 +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;
+	}
+}

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

@@ -151,7 +151,6 @@ 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 freedv_monitor;		// pass the Freedv audio to the speakers instead of to the demodulator
 static int softrock_correct_freq;	// gain and phase corrections for SoftRock
 static double softrock_correct_gain;
 static double softrock_correct_phase;
@@ -219,14 +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
@@ -569,15 +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
@@ -1851,10 +1848,8 @@ static int quisk_process_decimate(complex double * cSamples, int nSamples, int b
 static int quisk_process_demodulate(complex double * cSamples, double * dsamples, int nSamples, int bank, int nFilter, rx_mode_type rx_mode)
 {	// Changes here will require changes to get_filter_rate();
 	int i;
-	bool isLSB;
 	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 {
@@ -2090,6 +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);
@@ -2116,6 +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);
@@ -2154,102 +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. Modified by N2ADR 2024.
-		quisk_check_freedv_mode();
-		// current coding assumes input rate n_modem_sample_rate is 8000 or 48000.
-		// decimate:
-		switch (n_modem_sample_rate) {
-		case 8000:
-			quisk_filter_srate = quisk_decim_srate / 6;
-			nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to16, 3);
-			nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim16to8, 2);
-			break;
-		case 48000:
-			quisk_filter_srate = quisk_decim_srate;
-			break;
-		default:
-			quisk_filter_srate = quisk_decim_srate / 6;
-			QuiskPrintf("Bad FreeDV modem rate in quisk_process_demodulate\n");
-			break;
-		}
-		// filter and demodulate to mono audio:
-		isLSB = (rx_mode == FDV_L);
-		complex double cBuf[nSamples];
-		for (i = 0; i < nSamples; i++) {
-			cx = cRxFilterOut(cSamples[i], bank, nFilter);
-			if (isLSB)
-				dd = creal(cx) + cimag(cx);
-			else
-				dd = creal(cx) - cimag(cx);
-			if(bank == 0) {
-				measure_audio_sum += dd * dd;
-				measure_audio_count += 1;
-			}
-			cBuf[i] = dd;
-		}
-		// Perhaps decimate by an additional fraction
-		if (quisk_decim_srate != 48000) {
-			dd = quisk_decim_srate / 48000.0;
-			nSamples = cFracDecim(cBuf, nSamples, dd);
-			quisk_decim_srate = 48000;
-		}
-		if (bank == 0)
-			process_agc(&Agc1, cBuf, nSamples, 0);
-		else
-			process_agc(&Agc2, cBuf, nSamples, 0);
-		// pass data decimated down to n_modem_sample_rate
-		// return data is at n_speech_sample_rate
-#if 0
-		static double dmax_in = 0;
-		for (i = 0; i < nSamples; i++) {
-			dd = fabs(creal(cBuf[i])) / CLIP16;
-			if (dd > dmax_in)
-				dmax_in = dd;
-		}
-		if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
-			//printf("Max sample in %.0lf\n", dmax_in);
-			dmax_in = 0;
-		}
-#endif
-		if (nSamples > n_max_modem_samples) {
-			printf ("FreeDV: nSamples %d exceeds max_modem_samples %d\n", nSamples, n_max_modem_samples);
-			nSamples = n_max_modem_samples;
-		}
-		short sBuf[nSamples];
-		for (i = 0; i < nSamples; i++)
-			sBuf[i] = (short)(creal(cBuf[i]) / CLIP16);
-		if (freedv_monitor) {
-			for (i = 0; i < nSamples; i++)
-				dsamples[i] = (float)sBuf[i] * CLIP16;
-		}
-		else {
-			if (pt_quisk_freedv_rx)
-				nSamples = (* pt_quisk_freedv_rx)(sBuf, dsamples, nSamples, bank);
-		}
-#if 0
-		static double dmax_out = 0;
-		for (i = 0; i < nSamples; i++) {
-			dd = fabs(dsamples[i]) / CLIP16;
-			if (dd > dmax_out)
-				dmax_out = dd;
-		}
-		if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
-			//printf("      Max sample out %.0lf\n", dmax_out);
-			dmax_out = 0;
-		}
-#endif
-		// current coding assumes output rate n_speech_sample_rate is 8000 or 16000
-		switch (n_speech_sample_rate) {
-		case 8000:
-			nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
-			nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
-			break;
-		case 16000:
-			nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
-			break;
-		}
-		break;
 	}
 	if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
 		measured_audio = sqrt(measure_audio_sum / measure_audio_count) / CLIP32 * 1e6;
@@ -2544,7 +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.
@@ -2756,7 +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) {
@@ -2784,9 +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]);
@@ -2944,6 +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
@@ -2952,10 +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_modem_sample_rate;
-		break;
 	}
 	//QuiskPrintf("Filter rate %d\n", filter_srate);
 	return PyInt_FromLong(filter_srate);
@@ -3195,14 +3092,13 @@ 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", 
-	"freedv_monitor", NULL} ;
+	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, "|iiiiiiiidii", kwlist,
 	&quisk_is_vna, &nbytes, &py_sample_rx_endian, &read_error, &clip,
-	&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause,
-	&freedv_monitor))
+	&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause))
 		return NULL;
 	if (nbytes != -1) {
 		py_sample_rx_bytes = nbytes;
@@ -3243,31 +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;
 }
 
@@ -3815,19 +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;
@@ -3850,6 +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
@@ -6190,6 +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."},
@@ -6230,13 +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

@@ -312,10 +312,6 @@ 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 n_max_modem_samples;			// maximum input to freedv_rx()
 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
@@ -339,20 +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 *);
@@ -362,7 +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
@@ -396,7 +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);
@@ -404,12 +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)(short *, double *, int, int);
-typedef int  (* ty_dvoice_codec_tx)(complex double *, double *, int, 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 *);
@@ -466,7 +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])
@@ -481,7 +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.py

@@ -1,6 +1,6 @@
 #!/usr/bin/env python3
 
-# All QUISK software is Copyright (C) 2006-2024 by James C. Ahlstrom.
+# All QUISK software is Copyright (C) 2006-2025 by James C. Ahlstrom.
 # This free software is licensed for use under the GNU General Public
 # License (GPL), see http://www.opensource.org.
 # Note that there is NO WARRANTY AT ALL.  USE AT YOUR OWN RISK!!
@@ -292,9 +292,9 @@ class HamlibHandlerSerial:
 
   # Additional logic contributed by Dr. Karsten Schmidt
 
-  Mo2CoKen = {'CWL':7, 'CWU':3, 'LSB':1, 'USB':2, 'AM':5, 'FM':4, 'DGT-U':9, 'DGT-L':6, 'DGT-FM':4, 'DGT-IQ':9}
+  Mo2CoKen = {'CWL':7, 'CWU':3, 'LSB':1, 'USB':2, 'AM':5, 'FM':4, 'DGT-U':9, 'FDV-U':9, 'DGT-L':6, 'FDV-L':6, 'DGT-FM':4, 'DGT-IQ':9}
   Co2MoKen = {1:'LSB', 2:'USB', 3:'CWU', 4:'FM', 5:'AM', 6:'DGT-L', 7:'CWL', 9:'DGT-U'}
-  Mo2CoFlex = {'CWL':3, 'CWU':4, 'LSB':0, 'USB':1, 'AM':6, 'FM':5, 'DGT-U':7, 'DGT-L':9, 'DGT-FM':5, 'DGT-IQ':7}
+  Mo2CoFlex = {'CWL':3, 'CWU':4, 'LSB':0, 'USB':1, 'AM':6, 'FM':5, 'DGT-U':7, 'FDV-U':7, 'DGT-L':9, 'FDV-L':9, 'DGT-FM':5, 'DGT-IQ':7}
   Co2MoFlex = {0:'LSB', 1:'USB', 3:'CWL', 4:'CWU', 5:'FM', 6:'AM', 7:'DGT-U', 9:'DGT-L'}
   ConvertZZAC_P1_2_Step = { 0:     1,
                           1:    10,
@@ -585,11 +585,18 @@ class HamlibHandlerSerial:
     mode = self.app.mode
     info = cmd
     info += "%011d" % (self.app.rxFreq + self.app.VFO)	# frequency, ZZFA
-    info += '0000'
-    if ritFreq < 0:	# RIT freq
-      info += "-%05d" % -ritFreq
-    else:
-      info += "+%05d" % ritFreq
+    if len(cmd) == 4:	# Flex ZZIF
+      info += '0000'
+      if ritFreq < 0:	# RIT freq
+        info += "-%05d" % -ritFreq
+      else:
+        info += "+%05d" % ritFreq
+    else:		# Kenwood IF
+      info += '00000'
+      if ritFreq < 0:	# RIT freq
+        info += "-%04d" % -ritFreq
+      else:
+        info += "+%04d" % ritFreq
     info += "%d" % rit	# RIT status
     info += '0000'
     if QS.is_key_down():	# MOX, key down
@@ -933,6 +940,8 @@ class HamlibHandlerRig2:	# Test with telnet localhost 4532
       if not isinstance(text, Q3StringTypes):
         text = text.decode('utf-8')
       self.received += text
+      if HAMLIB_DEBUG and text:
+        print ("Raw rig2 received:", text)
     if '\n' in self.received:	# A complete command ending with newline is available
       self.input, self.received = self.received.split('\n', 1)	# Split off the command, save any further characters
     else:
@@ -986,17 +995,12 @@ class HamlibHandlerRig2:	# Test with telnet localhost 4532
     else:
       self.Send('0\n')
   def GetFreq(self):	# The Rx frequency
-    rx = self.app.multi_rx_screen.receiver_list
-    if rx:
-      rx = rx[0]
-      self.Reply('Frequency', rx.txFreq + rx.VFO, 0)
-      if HAMLIB_DEBUG:
-        print ("GetFreq rx", rx.txFreq, rx.VFO)
-    else:
-      self.Reply('Frequency', self.app.rxFreq + self.app.VFO, 0)
-      if HAMLIB_DEBUG:
-        print ("GetFreq app", self.app.rxFreq, self.app.txFreq, self.app.VFO)
+    # This is always the main Rx frequency without reference to added receivers.
+    self.Reply('Frequency', self.app.rxFreq + self.app.VFO, 0)
+    if HAMLIB_DEBUG:
+      print ("GetFreq app", self.app.rxFreq, self.app.txFreq, self.app.VFO)
   def SetFreq(self):	# The Rx frequency
+    # This is always the main Rx frequency without reference to added receivers.
     freq = self.GetParamNumber()
     try:
       freq = float(freq)
@@ -1005,12 +1009,7 @@ class HamlibHandlerRig2:	# Test with telnet localhost 4532
       self.ErrParam()
     else:
       freq = int(freq + 0.5)
-      # For multiple receivers this controls the first added receiver frequency; else it controls split Rx/Tx frequency.
-      rx = self.app.multi_rx_screen.receiver_list
-      if rx:
-        rx[0].ChangeRxTxFrequency(freq)
-      else:
-        self.app.ChangeRxTxFrequency(freq, None)
+      self.app.ChangeRxTxFrequency(freq, None)
   def GetSplitFreq(self):	# The Tx Frequency
     self.Reply('TX Frequency', self.app.txFreq + self.app.VFO, 0)
   def SetSplitFreq(self):	# The Tx Frequency
@@ -1052,9 +1051,9 @@ class HamlibHandlerRig2:	# Test with telnet localhost 4532
       mode = 'CW'
     elif mode == 'CWL':		# Is this what CWR means?
       mode = 'CWR'
-    elif mode == 'DGT-U':
+    elif mode in ('DGT-U', 'FDV-U'):
       mode = 'PKTUSB'
-    elif mode == 'DGT-L':
+    elif mode in ('DGT-L', 'FDV-L'):
       mode = 'PKTLSB'
     elif mode == 'DGT-FM':
       mode = 'PKTFM'
@@ -1218,6 +1217,267 @@ class HamlibHandlerRig2:	# Test with telnet localhost 4532
     else:
       self.ErrParam()
 
+class CatControl:
+  def GetFreqVfo(self, vfoB=False):
+    if vfoB:	# The Tx Frequency
+      return self.app.txFreq + self.app.VFO
+    else:	# The Rx Frequency
+      rx = self.app.multi_rx_screen.receiver_list
+      if rx:
+        rx = rx[0]
+        return rx.txFreq + rx.VFO
+      else:
+        return self.app.rxFreq + self.app.VFO
+  def SetFreqVfo(self, freq, vfoB=False):
+    if vfoB:	# The Tx Frequency
+      self.app.ChangeRxTxFrequency(None, freq)
+    else:	# The Rx Frequency
+      # For multiple receivers this controls the first added receiver frequency; else it controls split Rx/Tx frequency.
+      rx = self.app.multi_rx_screen.receiver_list
+      if rx:
+        rx[0].ChangeRxTxFrequency(freq)
+      else:
+        self.app.ChangeRxTxFrequency(freq, None)
+  def GetFilterBW(self):
+    return self.app.filter_bandwidth
+  def SetFilterBW(self, bw):
+    # Choose button closest to requested bandwidth
+    buttons = self.app.filterButns.GetButtons()
+    Lab = buttons[0].GetLabel()
+    diff = abs(int(Lab) - bw)
+    for i in range(1, len(buttons) - 1):
+      label = buttons[i].GetLabel()
+      df = abs(int(label) - bw)
+      if df < diff:
+        Lab = label
+        diff = df
+    self.app.filterButns.SetLabel(Lab, True)
+
+class ElecraftK4Handler(CatControl):	# Test with telnet localhost 9200
+  Mo2CoElecraft = {'LSB':1, 'USB':2, 'CWU':3, 'FM':4, 'AM':5, 'DGT-U':6, 'CWL':7, 'DGT-L':9, 'DGT-FM':4, 'DGT-IQ':6}
+  Co2MoElecraft = {1:'LSB', 2:'USB', 3:'CWU', 4:'FM', 5:'AM', 6:'DGT-U', 7:'CWL', 9:'DGT-L'}
+  def __init__(self, app, conf, sock, address):
+    self.app = app
+    self.conf = conf
+    self.sock = sock
+    sock.settimeout(0.0)
+    self.address = address
+    self.params = ''	# params is the string following the command
+    self.received = ''
+    self.cmds = []
+    self.K31Elecraft = False	# K31 mode
+    h = self.Handlers = {}	# Branch on the first two letters of the command
+    h['AI']	= self.AiAutoInfo
+    h['CW']	= self.CwCWPitch
+    h['DT']	= self.DtDataMode
+    h['FA']	= self.FaFbVfoFreq
+    h['FB']	= self.FaFbVfoFreq
+    h['FT']	= self.FtTxVfo
+    h['FW']	= self.FwFilterBW
+    h['ID']	= self.IdIdent
+    h['IF']	= self.IfInfo
+    h['IS']	= self.IsCenterPitch
+    h['KS']	= self.KsKeyerSpeed
+    h['K3']	= self.K3Elecraft
+    h['LN']	= self.LnLinkVfo
+    h['MD']	= self.MdMode
+    h['OM']	= self.OmOptionModules
+    h['RV']	= self.RvFirmware
+    h['RX']	= self.RxReceive
+    h['SB']	= self.SbSubReceiver
+    h['SM']	= self.SmSmeter
+    h['TX']	= self.TxTransmit
+  def Send(self, text):
+    """Send text back to the client."""
+    if HAMLIB_DEBUG:
+      for cmd in self.cmds[0:-1]:
+        print ("CatTcp received: %-10s" % cmd)
+      print ("CatTcp received: %-10s ; sent %s" % (self.cmds[-1], text))
+      del self.cmds[:]
+    text = text.encode('utf-8', errors='ignore')
+    try:
+      self.sock.sendall(text)
+    except socket.error:
+      self.sock.close()
+      self.sock = None
+  def Process(self):
+    """This is the main processing loop, and is called frequently.  It reads and satisfies requests."""
+    if not self.sock:
+      return 0
+    try:	# Read any data from the socket
+      text = self.sock.recv(1024)
+    except socket.timeout:	# This does not work
+      pass
+    except socket.error:	# Nothing to read
+      pass
+    else:					# We got some characters
+      text = text.decode('utf-8', errors='ignore')
+      self.received += text
+      if HAMLIB_DEBUG and text:
+        pass #print ("Raw CatTcp received:", text)
+    if ';' in self.received:	# A complete command ending with semicolon is available
+      cmd, self.received = self.received.split(';', 1)	# Split off the command, save any further characters
+    else:
+      return 1
+    cmd = cmd.strip()		# Here is our command line
+    if len(cmd) < 2:
+      return 1
+    if HAMLIB_DEBUG:
+      self.cmds.append(cmd)
+    base = cmd[0:2].upper()
+    try:
+      handler = self.Handlers[base]
+    except KeyError:
+      self.Send(cmd[0:2] + "?;")
+      #print ("Unknown command", cmd)
+      return 1
+    if len(cmd) == 2:
+      args = ''
+    elif cmd[2] == '$': # base is the first two letters upper case and any following "$"
+      base += '$'
+      args = cmd[3:]
+    else:
+      args = cmd[2:]
+    handler(base, args) # args is the remainder
+    return 1
+  def AiAutoInfo(self, base, args):
+    if not args:
+      self.Send("AI0;")
+    elif args != '0':
+      self.Send("AI0;")
+  def CwCWPitch(self, base, args):
+    pitch = self.conf.cwTone
+    pitch = (pitch + 5) // 10
+    if pitch < 25:
+      pitch = 25
+    elif pitch > 95:
+      pitch = 95
+    self.Send("CW%d;" % pitch)
+  def DtDataMode(self, base, args):
+    self.Send("%s0;" % base)
+  def IdIdent(self, base, args):
+    if args:
+      self.Send("ID?;")
+    else:
+      self.Send("ID017;")
+      #self.Send("IDQuisk;")
+  def IfInfo(self, base, args):
+    rxfreq = self.GetFreqVfo()
+    info = "%011d     " % rxfreq	# Rx frequency, five blanks
+    ritFreq = self.app.ritScale.GetValue()
+    if ritFreq < 0:	# +/- RIT freq
+      info += "-%04d" % -ritFreq
+    else:
+      info += "+%04d" % ritFreq
+    if self.app.ritButton.GetValue():	# RIT status, XIT status, blank "00"
+      info += "10 00"
+    else:
+      info += "00 00"
+    if QS.is_key_down():	# MOX, key down
+      info += '1'
+    else:
+      info += '0'
+    mode = self.Mo2CoElecraft.get(self.app.mode, 2)
+    info += "%d" % mode		# Mode 1-9
+    info += '00'		# Rx on VFOA, no scan 
+    if self.app.split_rxtx:	# VFO split status
+      info += '1'
+    else:
+      info += '0'
+    info += '001 ;'
+    self.Send(info)
+  def FaFbVfoFreq(self, base, args):
+    if base[1] in ('B', 'b'):
+      vfoB = True
+    else:
+      vfoB = False
+    if not args:	# Get command
+      freq = "%06d" % self.GetFreqVfo(vfoB)
+      self.Send(base + freq + ';')
+    else:		# Put command
+      try:
+        freq = int(args)
+      except:
+        self.Send(base + "?;")
+      else:
+        l = len(args)
+        if l <= 2:
+          freq *= 1000000
+        elif l <= 5:
+          freq *= 1000
+        self.SetFreqVfo(freq, vfoB)
+  def FtTxVfo(self, base, args):
+    if not args:
+      if self.app.split_rxtx:
+        self.Send("FT1;")
+      else:
+        self.Send("FT0;")
+    else:
+      # Set split Rx/Tx mode.
+      is_split = self.app.splitButton.GetValue()
+      split = int(args)
+      if split and not is_split:
+        self.app.splitButton.SetValue(split, True)
+      elif is_split and not split:
+        self.app.splitButton.SetValue(split, True)
+  def FwFilterBW(self, base, args):
+    if not args:
+      code = self.GetFilterBW()
+      self.Send("%s%04d;" % (base, (code + 5) // 10))
+    else:
+      self.SetFilterBW(int(args) * 10)
+  def IsCenterPitch(self, base, args):
+    if not args:
+      code = self.GetFilterBW() // 2
+      if self.K31Elecraft:
+        self.Send("%s %04d;" % (base, code))
+      else:
+        code = (code + 5) // 10
+        self.Send("%s%04d;" % (base, code))
+    else:
+      if self.K31Elecraft:
+        self.SetFilterBW(int(args) * 2)
+      else:
+        self.SetFilterBW(int(args) * 2 * 10)
+  def K3Elecraft(self, base, args):
+    if args == '1':
+      self.K31Elecraft = True
+    else:
+      self.K31Elecraft = False
+  def KsKeyerSpeed(self, base, args):
+    if len(base) == 2:
+      self.Send("KS013;")
+  def LnLinkVfo(self, base, args):
+    if args != '0':
+      self.Send("LN0;")
+  def MdMode(self, base, args):
+    if args:		# set the mode
+      code = int(args)
+      mode = self.Co2MoElecraft[code]
+      self.app.modeButns.SetLabel(mode, True)
+    else:		# return the mode
+      code = self.Mo2CoElecraft.get(self.app.mode, 2)
+      self.Send("%s%d;" % (base, code))
+  def OmOptionModules(self, base, args):
+    self.Send("OM ------------;")
+  def RvFirmware(self, base, args):
+    self.Send("%s99.99;" % base)
+  def RxReceive(self, base, args):
+    self.app.pttButton.SetValue(0, True)
+  def SbSubReceiver(self, base, args):
+    self.Send("SB0;")
+  def SmSmeter(self, base, args):
+    if base[0:3].upper() == "SMH":
+      self.Send(base + "?;")
+    elif args:
+      self.Send(base + "?;")
+    elif self.K31Elecraft:
+      self.Send("SM0000;")
+    else:
+      self.Send("SM00;")
+  def TxTransmit(self, base, args):
+    self.app.pttButton.SetValue(1, True)
+
 class SoundThread(threading.Thread):
   """Create a second (non-GUI) thread to read, process and play sound."""
   def __init__(self, samples_from_python):
@@ -3024,16 +3284,27 @@ class MultiReceiverScreen(wx.SplitterWindow):
       if pane != excpt:
         pane.play_btn.SetValue(False)
         pane.is_playing = False
-  def OnAddReceiver(self, event):
+  def OnAddReceiver(self, event=None, hide_it=False):
     index = len(self.receiver_list)
     if index >= 7:
       return
     if index == 0:
-      pane2 = self.rx_zero
-      splitter = pane2.GetParent()
       pane1 = MultiRxGraph(self, self.data_width, self.graph_width, index)
       self.receiver_list.append(pane1)
-      splitter.SplitHorizontally(pane1, self.rx_zero)
+      if hide_it:
+        pane1.Show(False)
+        pane1.graph_display.Show(False)
+      else:
+        pane2 = self.rx_zero
+        splitter = pane2.GetParent()
+        splitter.SplitHorizontally(pane1, self.rx_zero)
+    elif index == 1 and not self.receiver_list[0].IsShown():
+        pane1 = self.receiver_list[0]
+        pane2 = self.rx_zero
+        splitter = pane2.GetParent()
+        pane1.Show(True)
+        pane1.graph_display.Show(True)
+        splitter.SplitHorizontally(pane1, self.rx_zero)
     else:
       pane2 = self.receiver_list[-1]
       parent = pane2.GetParent()
@@ -3446,8 +3717,8 @@ class App(wx.App):
   'txAudioPreemphUsb', 'txAudioPreemphAm', 'txAudioPreemphFm', 'txAudioPreemphFdv',
   'wfallScaleZ', 'wfallGrScaleZ', 'graphScaleZ', 'split_rxtx_play', 'modeFilter',
   'file_name_rec_audio', 'file_name_rec_samples', 'file_name_rec_mic', 'file_name_rec_tmp',
-  'file_name_play_audio', 'file_name_play_samples', 'file_name_play_cq', 'freedv_mode',
-  'file_play_source', 'hermes_LNA_dB']
+  'file_name_play_audio', 'file_name_play_samples', 'file_name_play_cq',
+  'file_play_source', 'hermes_LNA_dB', 'hermes_atten_dB']
   def __init__(self):
     global application
     QS.AppStatus(1)
@@ -3667,9 +3938,8 @@ class App(wx.App):
     self.hot_key_ptt_active = False
     self.serial_ptt_active = False
     self.vox_ptt_active = False
-    self.freedv_mode = 'Mode 700D'		# restore FreeDV mode setting
-    self.freedv_menu = None
     self.hermes_LNA_dB = 20
+    self.hermes_atten_dB = 0
     if hasattr(Hardware, "OnChangeRxTx"):
       self.want_RxTx = True
     else:
@@ -3698,6 +3968,19 @@ class App(wx.App):
         # traceback.print_exc()
     else:
       self.hamlib_socket = None
+    # Quisk control by K4 commands through a TCP port
+    self.k4_tcp_clients = []	# list of TCP connections to handle
+    if conf.k4_tcp_port:
+      try:
+        self.k4_tcp_socket = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
+        self.k4_tcp_socket.bind((conf.k4_tcp_ip, conf.k4_tcp_port))
+        self.k4_tcp_socket.settimeout(0.0)
+        self.k4_tcp_socket.listen(5)	# listen for TCP connections from multiple clients
+      except:
+        self.k4_tcp_socket = None
+        # traceback.print_exc()
+    else:
+      self.k4_tcp_socket = None
     # Quisk control by fldigi
     self.fldigi_new_freq = None
     self.fldigi_freq = None
@@ -3925,6 +4208,8 @@ class App(wx.App):
     # Start wdsp
     self.wdsp = quisk_wdsp.Cwdsp(self)
     self.wdsp_channel = 1
+    self.wdsp_SNB = 0
+    self.wdsp_NR2 = 0
     self.wdsp.open(self.wdsp_channel)
     #print ('data_width %d, FFT size %d, FFT mult %d, average_count %d, rate %d, Refresh %.2f Hz' % (
     #    self.data_width, self.fft_size, self.fft_size / self.data_width, average_count, self.sample_rate,
@@ -3990,6 +4275,10 @@ class App(wx.App):
     self.main_frame.SetClientSize(wx.Size(width, self.height))
     if hasattr(Hardware, 'pre_open'):       # pre_open() is called before open()
       Hardware.pre_open()
+    try:
+      self.is_HermesLite2 = Hardware.is_HermesLite2	# This is set in pre_open()
+    except:
+      self.is_HermesLite2 = False
     if self.local_conf.GetWidgets(self, Hardware, conf, frame, gbs, vertBox):
       pass
     elif conf.quisk_widgets:
@@ -4406,14 +4695,15 @@ class App(wx.App):
     if self.hamlib_socket:
       self.hamlib_socket.close()
       self.hamlib_socket = None
+    if self.k4_tcp_socket:
+      self.k4_tcp_socket.close()
+      self.k4_tcp_socket = None
     if self.dxCluster:
       self.dxCluster.stop()
     if self.hamlib_com1_handler:
       self.hamlib_com1_handler.close()
     if self.hamlib_com2_handler:
       self.hamlib_com2_handler.close()
-    if conf.add_freedv_button:
-      QS.freedv_close()
     msg = QS.GetQuiskPrintf()
     if msg:
       print(msg, end='')
@@ -4611,7 +4901,7 @@ class App(wx.App):
     b.char_shortcut = 'u'
     self.MakeAccel(b)
     left_row2.append(b)
-    b = QuiskCheckbutton(frame, self.OnBtnNR2, text='NR2')
+    b = QuiskCycleCheckbutton(frame, self.OnBtnNR2, ['NR2', 'NR2', 'NR2t'])
     if not self.wdsp.version:
       b.Enable(False)
     left_row2.append(b)
@@ -4752,14 +5042,9 @@ class App(wx.App):
     self.config_screen.config.EnableRecPlay()
     ### Right bank of buttons
     mode_names = ['CWL', 'CWU', 'LSB', 'USB', 'AM', 'FM', 'DGT-U', 'DGT-L', 'DGT-FM', 'DGT-IQ', 'FDV-U', 'FDV-L', 'IMD']
-    labels = [('CWL', 'CWU'), ('LSB', 'USB'), 'AM', 'FM', ('DGT-U', 'DGT-L', 'DGT-FM', 'DGT-IQ')]
-    shortcuts = ['C', 'S', 'A', 'M', 'D']
-    count = 5	# There is room for seven buttons
-    if conf.add_freedv_button:
-      n_freedv = count
-      count += 1
-      labels.append('FDV-U')
-      shortcuts.append('F')
+    labels = [('CWL', 'CWU'), ('LSB', 'USB'), 'AM', 'FM', ('DGT-U', 'DGT-L', 'DGT-FM', 'DGT-IQ'), ('FDV-U', 'FDV-L')]
+    shortcuts = ['C', 'S', 'A', 'M', 'D', 'F']
+    count = 6	# There is room for seven buttons
     if conf.add_imd_button:
       n_imd = count
       count += 1
@@ -4781,45 +5066,13 @@ class App(wx.App):
       self.modeButns.buttons[0].idName = "CW U/L"
       self.modeButns.buttons[1].idName = "SSB U/L"
       self.modeButns.buttons[4].idName = "DGT"
+      self.modeButns.buttons[5].idName = "FDV"
     else:
       labels = ['CWL', 'CWU', 'LSB', 'USB', 'AM', 'FM', 'DGT-U', 'DGT-L', 'DGT-FM', 'DGT-IQ', 'FDV-U', 'IMD']
       self.modeButns = RadioBtnPopup(frame, self.OnBtnMode, labels, None, 'modeButns')
     self.modeButns.idName = "modeButns"
-    self.freedv_menu_items = {}
-    if conf.add_freedv_button:
-      self.freedv_menu = QuiskMenu("freedv_menu")
-      msg = conf.freedv_tx_msg
-      QS.freedv_set_options(mode=conf.freedv_modes[0][1], tx_msg=msg, DEBUG=0, squelch=1)
-      self.freedv_menu.AppendCheckItem("Monitor", self.OnFreedvMonitor)
-      self.freedv_menu.AppendSeparator()
-      for mode, index in conf.freedv_modes:
-        item = self.freedv_menu.AppendRadioItem(mode, self.OnFreedvMenu, mode == self.freedv_mode)
-        self.freedv_menu_items[index] = item
-        if mode == self.freedv_mode:	# Restore mode
-          QS.freedv_set_options(mode=index)
-        if mode in ("Mode 2020", "Mode 2400A", "Mode 2400B"):
-          item.Enable(False)
-      if conf.button_layout == 'Large screen':
-        b = QuiskCycleCheckbutton(frame, None, ('FDV-U', 'FDV-L'), is_radio=True)
-        b.idName = "FDV"
-        b.char_shortcut = 'F'
-        self.btnFreeDV = WrapMenu(b, self.freedv_menu)
-        self.modeButns.ReplaceButton(n_freedv, self.btnFreeDV)
-      else:
-        self.btnFreeDV = self.modeButns.AddMenu('FDV-U', self.freedv_menu)
-      try:
-        ok = QS.freedv_open()
-      except:
-        traceback.print_exc()
-        ok = 0
-      if not ok:
-        conf.add_freedv_button = False
-        if conf.button_layout == 'Large screen':
-          self.modeButns.GetButtons()[n_freedv].Enable(0)
-        else:
-          self.modeButns.Enable('FDV-U', False)
     if conf.add_imd_button:
-      val = 500
+      val = 1000
       QS.set_imd_level(val)
       if conf.button_layout == 'Large screen':
         b = QuiskCheckbutton(frame, None, 'IMD', color=conf.color_test)
@@ -5096,16 +5349,6 @@ class App(wx.App):
     t = '%13.2f' % (QS.measure_frequency(-1) + vfo)
     t = t[0:4] + ' ' + t[4:7] + ' ' + t[7:] + ' Hz'
     self.smeter.SetLabel(t)
-  def NewDVmeter(self):
-    if conf.add_freedv_button:
-      snr = QS.freedv_get_snr()
-      txt = QS.freedv_get_rx_char()
-      self.graph.ScrollMsg(txt)
-      self.waterfall.ScrollMsg(txt)
-    else:
-      snr = 0.0
-    t = "  SNR %3.0f" % snr
-    self.smeter.SetLabel(t)
   def NewSmeter(self):
     self.smeter_db_count += 1		# count for average
     x = QS.get_smeter()
@@ -5275,66 +5518,6 @@ class App(wx.App):
     self.multi_rx_screen.waterfall.pane2.filter_center = center
     if self.screen is self.filter_screen:
       self.screen.NewFilter()
-  def OnFreedvMonitor(self, event):
-    for item in self.freedv_menu.GetMenuItems():
-      if item.GetKind() == wx.ITEM_CHECK:
-        if item.IsChecked():
-          QS.set_params(freedv_monitor=1)
-        else:
-          QS.set_params(freedv_monitor=0)
-        break
-  def OnFreedvMenu(self, event):
-    text = ''
-    for item in self.freedv_menu.GetMenuItems():
-      kind = item.GetKind()
-      if kind != wx.ITEM_RADIO:
-        continue
-      if item.IsChecked():
-        text = item.GetItemLabel()
-        break
-    for mode, index in conf.freedv_modes:
-      if mode == text:
-        break
-    else:
-      print ("Failure in OnFreedvMenu")
-      return
-    mode = QS.freedv_set_options(mode=index)
-    if mode == index:
-      self.freedv_mode = text
-    elif not self.remote_control_slave:		# change to new mode failed
-      self.freedv_menu_items[mode].Check(1)
-      pos = (self.width//2, self.height//2)
-      if index == 8:
-        dlg = wx.MessageDialog(self.main_frame, "Quisk does not install %s. Please install a system-wide codec2." % text, "FreeDV Modes", wx.OK, pos)
-      else:
-        dlg = wx.MessageDialog(self.main_frame, "No codec2 support for " + text, "FreeDV Modes", wx.OK, pos)
-      dlg.ShowModal()
-    else:
-      self.freedv_menu_items[mode].Check(1)
-      print ("FreeDV change mode failed.")
-    self.OnChangeFilter()
-  def OnChangeFilter(self):
-    # Used by OnFreedvMenu because the FreeDV mode may change the filter sample rate
-    bw = int(self.filterButns.GetLabel())
-    # We can not use QS.get_filter_srate() because the FreeDV mode is not changed yet.
-    if self.freedv_mode in ("Mode 2400A", "Mode 2400B"):
-      frate = 48000;
-    else:
-      frate = 8000;
-    bw = min(bw, frate // 2)
-    self.filter_bandwidth = bw
-    center = self.GetFilterCenter(self.mode, bw)
-    filtI, filtQ = self.MakeFilterCoef(frate, None, bw, center)
-    lower_edge = center - bw // 2
-    QS.set_filters(filtI, filtQ, bw, lower_edge, 0)
-    self.multi_rx_screen.graph.filter_bandwidth = bw
-    self.multi_rx_screen.graph.filter_center = center
-    self.multi_rx_screen.waterfall.pane1.filter_bandwidth = bw
-    self.multi_rx_screen.waterfall.pane1.filter_center = center
-    self.multi_rx_screen.waterfall.pane2.filter_bandwidth = bw
-    self.multi_rx_screen.waterfall.pane2.filter_center = center
-    if self.screen is self.filter_screen:
-      self.screen.NewFilter()
   def OnBtnHelp(self, event):
     if event.GetEventObject().GetValue():
       self.OnBtnScreen(None, 'Help')
@@ -5799,11 +5982,6 @@ class App(wx.App):
         QS.set_squelch(self.levelSquelch / 12.0 - 120.0)
       else:
         QS.set_squelch(-999.0)
-    elif self.mode[0:3]  == 'FDV':
-      if value:
-        QS.freedv_set_squelch_en(1)
-      else:
-        QS.freedv_set_squelch_en(0)
     else:
       self.levelSquelchSSB = btn.slider_value
       if value:
@@ -5817,11 +5995,18 @@ class App(wx.App):
       QS.set_auto_notch(0)
   def OnBtnNR2(self, event):
     btn = event.GetEventObject()
-    if btn.GetValue():
+    if btn.index == 1:
+      self.wdsp.put(self.wdsp.SetRXAEMNRgainMethod, self.wdsp_channel, 2)
       self.wdsp.put(self.wdsp.SetRXAEMNRRun, self.wdsp_channel, 1)
-      QS.wdsp_set_parameter(self.wdsp_channel, in_use=1)
+      self.wdsp_NR2 = 1
+    elif btn.index == 2:
+      self.wdsp.put(self.wdsp.SetRXAEMNRgainMethod, self.wdsp_channel, 3)
+      self.wdsp.put(self.wdsp.SetRXAEMNRRun, self.wdsp_channel, 1)
+      self.wdsp_NR2 = 1
     else:
       self.wdsp.put(self.wdsp.SetRXAEMNRRun, self.wdsp_channel, 0)
+      self.wdsp_NR2 = 0
+    QS.wdsp_set_parameter(self.wdsp_channel, in_use=self.wdsp_NR2 + self.wdsp_SNB)
   def OnMenuNB(self, event):
     # menu is ("NB 1", "NB 2", "NB 3", "SNB")
     for item in self.NB_menu.GetMenuItems():
@@ -5837,14 +6022,17 @@ class App(wx.App):
       if text == "SNB":
         QS.set_noise_blanker(0)
         self.wdsp.put(self.wdsp.SetRXASNBARun, self.wdsp_channel, 1)
-        QS.wdsp_set_parameter(self.wdsp_channel, in_use=1)
+        self.wdsp_SNB = 1
       else:
         self.wdsp.put(self.wdsp.SetRXASNBARun, self.wdsp_channel, 0)
+        self.wdsp_SNB = 0
         index = int(text[-1])
         QS.set_noise_blanker(index)
     else:	# Noise blanker Off
       QS.set_noise_blanker(0)
       self.wdsp.put(self.wdsp.SetRXASNBARun, self.wdsp_channel, 0)
+      self.wdsp_SNB = 0
+    QS.wdsp_set_parameter(self.wdsp_channel, in_use=self.wdsp_NR2 + self.wdsp_SNB)
   def FreqEntry(self, event):
     freq = event.GetString()
     win = event.GetEventObject()
@@ -6383,7 +6571,7 @@ class App(wx.App):
         if rxtx != 'rx':
           self.fldigi_server.main.rx()
           self.fldigi_timer = time.time()
-  def HamlibPoll(self):		# Poll for Hamlib commands
+  def HamlibPoll(self):		# Poll for Hamlib and K4 TCP commands
     if self.hamlib_socket:
       try:		# Poll for new client connections.
         conn, address = self.hamlib_socket.accept()
@@ -6397,6 +6585,19 @@ class App(wx.App):
           self.hamlib_clients.remove(client)
           # print ('Remove', client.address)
           break
+    if self.k4_tcp_socket:
+      try:		# Poll for new client connections.
+        conn, address = self.k4_tcp_socket.accept()
+      except socket.error:
+        pass
+      else:
+        # print ('Connection from', address)
+        self.k4_tcp_clients.append(ElecraftK4Handler(self, conf, conn, address))
+      for client in self.k4_tcp_clients:	# Service existing clients
+        if not client.Process():		# False return indicates a closed connection; remove the handler for this client
+          self.k4_tcp_clients.remove(client)
+          # print ('Remove', client.address)
+          break
   def OnKeyHook(self, event):
     event.Skip()
     if conf.hot_key_ptt_if_hidden:	# Hot key PTT operates even if Quisk is hidden
@@ -6574,10 +6775,7 @@ class App(wx.App):
             d = 1E-10
           self.smeter.SetLabel(" ADC %.0f%% %.0fdB" % (d * 100.0, 20 * math.log10(d)))
         elif self.smeter_usage == "smeter":		# update the S-meter
-          if self.mode in ('FDV-U', 'FDV-L'):
-            self.NewDVmeter()
-          else:
-            self.NewSmeter()
+          self.NewSmeter()
         elif self.smeter_usage == "freq":
           self.MeasureFrequency()	# display measured frequency
         else:
@@ -6605,7 +6803,7 @@ class App(wx.App):
     if self.timer - self.heart_time0 > 0.10:	# call hardware to perform background tasks:
       self.heart_time0 = self.timer
       Hardware.HeartBeat()
-      if conf.hermes_lite2_enable:
+      if self.is_HermesLite2:
         self.tx_inhibit = QS.get_params('quisk_tx_inhibit')
       else:
         self.tx_inhibit = 0

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
 
@@ -473,13 +478,6 @@ hermes_iob_rxin = 'J10 available'
 #hermes_iob_rxin = 'HL2 Rx to J9'
 #hermes_iob_rxin = 'Use J9 and J10'
 
-## hermes_lite2_enable			Hermes Lite2 protocol, boolean
-# This turns on some of the protocol extensions for the Hermes Lite2 that are not part
-# of the official Hermes Protocol 1. Currently this only enables Tx Inhibit (CN8).
-# Changes are immediate (no need to restart).
-hermes_lite2_enable = True
-#hermes_lite2_enable = False
-
 
 # These are known power meter calibration tables. This table is not present in the JSON settings file.
 power_meter_std_calibrations = {}
@@ -490,6 +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.
@@ -507,17 +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
@@ -585,7 +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'
@@ -601,7 +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'
@@ -632,7 +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'
@@ -743,7 +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'
@@ -758,7 +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
@@ -827,7 +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'
@@ -859,7 +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.
@@ -1094,6 +1209,11 @@ 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:
 #
 # In PortAudio, soundcards have an index number 0, 1, 2, ... and a name.
@@ -1303,51 +1423,15 @@ fixed_tune_offset = 0
 
 
 ################ 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.
@@ -1398,19 +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 = ""
@@ -1782,23 +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 700E', 13), ('Mode 2020', 8))
-
 # 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)

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."""

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 = '4.2.41'
+Version = '4.2.50'
 
 fp = open("__init__.py", "w")	# write title string
 fp.write("#Quisk version %s\n" % Version)

sound.c

@@ -1000,21 +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, 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

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
+authors' reputations.
+
+  Finally, any free program is threatened constantly by software
+patents.  We wish to avoid the danger that redistributors of a free
+program will individually obtain patent licenses, in effect making the
+program proprietary.  To prevent this, we have made it clear that any
+patent must be licensed for everyone's free use or not licensed at all.
+
+  The precise terms and conditions for copying, distribution and
+modification follow.
+
+                    GNU GENERAL PUBLIC LICENSE
+   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
+
+  0. This License applies to any program or other work which contains
+a notice placed by the copyright holder saying it may be distributed
+under the terms of this General Public License.  The "Program", below,
+refers to any such program or work, and a "work based on the Program"
+means either the Program or any derivative work under copyright law:
+that is to say, a work containing the Program or a portion of it,
+either verbatim or with modifications and/or translated into another
+language.  (Hereinafter, translation is included without limitation in
+the term "modification".)  Each licensee is addressed as "you".
+
+Activities other than copying, distribution and modification are not
+covered by this License; they are outside its scope.  The act of
+running the Program is not restricted, and the output from the Program
+is covered only if its contents constitute a work based on the
+Program (independent of having been made by running the Program).
+Whether that is true depends on what the Program does.
+
+  1. You may copy and distribute verbatim copies of the Program's
+source code as you receive it, in any medium, provided that you
+conspicuously and appropriately publish on each copy an appropriate
+copyright notice and disclaimer of warranty; keep intact all the
+notices that refer to this License and to the absence of any warranty;
+and give any other recipients of the Program a copy of this License
+along with the Program.
+
+You may charge a fee for the physical act of transferring a copy, and
+you may at your option offer warranty protection in exchange for a fee.
+
+  2. You may modify your copy or copies of the Program or any portion
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+distribute such modifications or work under the terms of Section 1
+above, provided that you also meet all of these conditions:
+
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+    stating that you changed the files and the date of any change.
+
+    b) You must cause any work that you distribute or publish, that in
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+    part thereof, to be licensed as a whole at no charge to all third
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+
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+    when run, you must cause it, when started running for such
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+
+These requirements apply to the modified work as a whole.  If
+identifiable sections of that work are not derived from the Program,
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+distribute the same sections as part of a whole which is a work based
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+
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+
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+with the Program (or with a work based on the Program) on a volume of
+a storage or distribution medium does not bring the other work under
+the scope of this License.
+
+  3. You may copy and distribute the Program (or a work based on it,
+under Section 2) in object code or executable form under the terms of
+Sections 1 and 2 above provided that you also do one of the following:
+
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+
+    c) Accompany it with the information you received as to the offer
+    to distribute corresponding source code.  (This alternative is
+    allowed only for noncommercial distribution and only if you
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+    an offer, in accord with Subsection b above.)
+
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+
+If distribution of executable or object code is made by offering
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+access to copy the source code from the same place counts as
+distribution of the source code, even though third parties are not
+compelled to copy the source along with the object code.
+
+  4. You may not copy, modify, sublicense, or distribute the Program
+except as expressly provided under this License.  Any attempt
+otherwise to copy, modify, sublicense or distribute the Program is
+void, and will automatically terminate your rights under this License.
+However, parties who have received copies, or rights, from you under
+this License will not have their licenses terminated so long as such
+parties remain in full compliance.
+
+  5. You are not required to accept this License, since you have not
+signed it.  However, nothing else grants you permission to modify or
+distribute the Program or its derivative works.  These actions are
+prohibited by law if you do not accept this License.  Therefore, by
+modifying or distributing the Program (or any work based on the
+Program), you indicate your acceptance of this License to do so, and
+all its terms and conditions for copying, distributing or modifying
+the Program or works based on it.
+
+  6. Each time you redistribute the Program (or any work based on the
+Program), the recipient automatically receives a license from the
+original licensor to copy, distribute or modify the Program subject to
+these terms and conditions.  You may not impose any further
+restrictions on the recipients' exercise of the rights granted herein.
+You are not responsible for enforcing compliance by third parties to
+this License.
+
+  7. If, as a consequence of a court judgment or allegation of patent
+infringement or for any other reason (not limited to patent issues),
+conditions are imposed on you (whether by court order, agreement or
+otherwise) that contradict the conditions of this License, they do not
+excuse you from the conditions of this License.  If you cannot
+distribute so as to satisfy simultaneously your obligations under this
+License and any other pertinent obligations, then as a consequence you
+may not distribute the Program at all.  For example, if a patent
+license would not permit royalty-free redistribution of the Program by
+all those who receive copies directly or indirectly through you, then
+the only way you could satisfy both it and this License would be to
+refrain entirely from distribution of the Program.
+
+If any portion of this section is held invalid or unenforceable under
+any particular circumstance, the balance of the section is intended to
+apply and the section as a whole is intended to apply in other
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+
+It is not the purpose of this section to induce you to infringe any
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+such claims; this section has the sole purpose of protecting the
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+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
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+make exceptions for this.  Our decision will be guided by the two goals
+of preserving the free status of all derivatives of our free software and
+of promoting the sharing and reuse of software generally.
+
+                            NO WARRANTY
+
+  11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY
+FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW.  EXCEPT WHEN
+OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES
+PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED
+OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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+TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU.  SHOULD THE
+PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,
+REPAIR OR CORRECTION.
+
+  12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR
+REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,
+INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING
+OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED
+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
+delay.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+delay.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h
+delay.o: fmd.h iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h
+delay.o: iqc.h main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h
+delay.o: patchpanel.h resample.h rmatch.h varsamp.h RXA.h sender.h shift.h
+delay.o: siphon.h slew.h snb.h ssql.h syncbuffs.h TXA.h utilities.h
+dexp.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+dexp.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+dexp.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+dexp.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+dexp.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+dexp.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+dexp.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+div.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+div.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+div.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+div.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+div.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+div.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+div.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+eer.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h firmin.h
+eer.o: calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h compress.h
+eer.o: dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h iir.h
+eer.o: wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h main.h
+eer.o: meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+eer.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
+eer.o: snb.h ssql.h syncbuffs.h TXA.h utilities.h
+emnr.o: comm.h amd.h ammod.h amsq.h analyzer.h anf.h anr.h bandpass.h
+emnr.o: firmin.h calcc.h delay.h lmath.h cblock.h cfcomp.h cfir.h channel.h
+emnr.o: compress.h dexp.h div.h eer.h emnr.h emph.h eq.h fcurve.h fir.h fmd.h
+emnr.o: iir.h wcpAGC.h fmmod.h fmsq.h gain.h gen.h icfir.h iobuffs.h iqc.h
+emnr.o: main.h meter.h meterlog10.h nbp.h nob.h nobII.h osctrl.h patchpanel.h
+emnr.o: resample.h rmatch.h varsamp.h RXA.h sender.h shift.h siphon.h slew.h
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+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;
+}

wdsp/fcurve.h

@@ -0,0 +1,34 @@
+/*  fcurve.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
+
+*/
+
+#ifndef _fcurve_h
+#define _fcurve_h
+
+extern double* fc_impulse (int nc, double f0, double f1, double g0, double g1, int curve, double samplerate, double scale, int ctfmode, int wintype);
+
+extern double* fc_mults (int size, double f0, double f1, double g0, double g1, int curve, double samplerate, double scale, int ctfmode, int wintype);
+
+#endif

wdsp/fir.c

@@ -0,0 +1,401 @@
+/*  fir.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2016, 2022 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@pratt.one
+*/
+
+#define _CRT_SECURE_NO_WARNINGS
+#include "comm.h"
+
+double* fftcv_mults (int NM, double* c_impulse)
+{
+    double* mults        = (double *) malloc0 (NM * sizeof (complex));
+    double* cfft_impulse = (double *) malloc0 (NM * sizeof (complex));
+    fftw_plan ptmp = fftw_plan_dft_1d(NM, (fftw_complex *) cfft_impulse,
+            (fftw_complex *) mults, FFTW_FORWARD, FFTW_PATIENT);
+    memset (cfft_impulse, 0, NM * sizeof (complex));
+    // store complex coefs right-justified in the buffer
+    memcpy (&(cfft_impulse[NM - 2]), c_impulse, (NM / 2 + 1) * sizeof(complex));
+    fftw_execute (ptmp);
+    fftw_destroy_plan (ptmp);
+    _aligned_free (cfft_impulse);
+    return mults;
+}
+
+double* get_fsamp_window(int N, int wintype)
+{
+    int i;
+    double arg0, arg1;
+    double* window = (double *) malloc0 (N * sizeof(double));
+    switch (wintype)
+    {
+    case 0:
+        arg0 = 2.0 * PI / ((double)N - 1.0);
+        for (i = 0; i < N; i++)
+        {
+            arg1 = cos(arg0 * (double)i);
+            window[i]  =   +0.21747
+                + arg1 *  (-0.45325
+                + arg1 *  (+0.28256
+                + arg1 *  (-0.04672)));
+        }
+        break;
+    case 1:
+        arg0 = 2.0 * PI / ((double)N - 1.0);
+        for (i = 0; i < N; ++i)
+        {
+            arg1 = cos(arg0 * (double)i);
+            window[i]  =   +6.3964424114390378e-02
+                + arg1 *  (-2.3993864599352804e-01
+                + arg1 *  (+3.5015956323820469e-01
+                + arg1 *  (-2.4774111897080783e-01
+                + arg1 *  (+8.5438256055858031e-02
+                + arg1 *  (-1.2320203369293225e-02
+                + arg1 *  (+4.3778825791773474e-04))))));
+        }
+        break;
+    default:
+        for (i = 0; i < N; i++)
+            window[i] = 1.0;
+    }
+    return window;
+}
+
+double* fir_fsamp_odd (int N, double* A, int rtype, double scale, int wintype)
+{
+    int i, j;
+    int mid = (N - 1) / 2;
+    double mag, phs;
+    double* window;
+    double *fcoef     = (double *) malloc0 (N * sizeof (complex));
+    double *c_impulse = (double *) malloc0 (N * sizeof (complex));
+    fftw_plan ptmp = fftw_plan_dft_1d(N, (fftw_complex *)fcoef, (fftw_complex *)c_impulse, FFTW_BACKWARD, FFTW_PATIENT);
+    double local_scale = 1.0 / (double)N;
+    for (i = 0; i <= mid; i++)
+    {
+        mag = A[i] * local_scale;
+        phs = - (double)mid * TWOPI * (double)i / (double)N;
+        fcoef[2 * i + 0] = mag * cos (phs);
+        fcoef[2 * i + 1] = mag * sin (phs);
+    }
+    for (i = mid + 1, j = 0; i < N; i++, j++)
+    {
+        fcoef[2 * i + 0] = + fcoef[2 * (mid - j) + 0];
+        fcoef[2 * i + 1] = - fcoef[2 * (mid - j) + 1];
+    }
+    fftw_execute (ptmp);
+    fftw_destroy_plan (ptmp);
+    _aligned_free (fcoef);
+    window = get_fsamp_window(N, wintype);
+    switch (rtype)
+    {
+    case 0:
+        for (i = 0; i < N; i++)
+            c_impulse[i] = scale * c_impulse[2 * i] * window[i];
+        break;
+    case 1:
+        for (i = 0; i < N; i++)
+        {
+            c_impulse[2 * i + 0] *= scale * window[i];
+            c_impulse[2 * i + 1] = 0.0;
+        }
+        break;
+    }
+    _aligned_free (window);
+    return c_impulse;
+}
+
+double* fir_fsamp (int N, double* A, int rtype, double scale, int wintype)
+{
+    int n, i, j, k;
+    double sum;
+    double* window;
+    double *c_impulse = (double *) malloc0 (N * sizeof (complex));
+
+    if (N & 1)
+    {
+        int M = (N - 1) / 2;
+        for (n = 0; n < M + 1; n++)
+        {
+            sum = 0.0;
+            for (k = 1; k < M + 1; k++)
+                sum += 2.0 * A[k] * cos(TWOPI * (n - M) * k / N);
+            c_impulse[2 * n + 0] = (1.0 / N) * (A[0] + sum);
+            c_impulse[2 * n + 1] = 0.0;
+        }
+        for (n = M + 1, j = 1; n < N; n++, j++)
+        {
+            c_impulse[2 * n + 0] = c_impulse[2 * (M - j) + 0];
+            c_impulse[2 * n + 1] = 0.0;
+        }
+    }
+    else
+    {
+        double M = (double)(N - 1) / 2.0;
+        for (n = 0; n < N / 2; n++)
+        {
+            sum = 0.0;
+            for (k = 1; k < N / 2; k++)
+                sum += 2.0 * A[k] * cos(TWOPI * (n - M) * k / N);
+            c_impulse[2 * n + 0] = (1.0 / N) * (A[0] + sum);
+            c_impulse[2 * n + 1] = 0.0;
+        }
+        for (n = N / 2, j = 1; n < N; n++, j++)
+        {
+            c_impulse[2 * n + 0] = c_impulse[2 * (N / 2 - j) + 0];
+            c_impulse[2 * n + 1] = 0.0;
+        }
+    }
+    window = get_fsamp_window (N, wintype);
+    switch (rtype)
+    {
+    case 0:
+        for (i = 0; i < N; i++)
+            c_impulse[i] = scale * c_impulse[2 * i] * window[i];
+        break;
+    case 1:
+        for (i = 0; i < N; i++)
+            {
+                c_impulse[2 * i + 0] *= scale * window[i];
+                c_impulse[2 * i + 1] = 0.0;
+            }
+        break;
+    }
+    _aligned_free (window);
+    return c_impulse;
+}
+
+double* fir_bandpass (int N, double f_low, double f_high, double samplerate, int wintype, int rtype, double scale)
+{
+    double *c_impulse = (double *) malloc0 (N * sizeof (complex));
+    double ft = (f_high - f_low) / (2.0 * samplerate);
+    double ft_rad = TWOPI * ft;
+    double w_osc = PI * (f_high + f_low) / samplerate;
+    int i, j;
+    double m = 0.5 * (double)(N - 1);
+    double delta = PI / m;
+    double cosphi;
+    double posi, posj;
+    double sinc, window, coef;
+
+    if (N & 1)
+    {
+        switch (rtype)
+        {
+        case 0:
+            c_impulse[N >> 1] = scale * 2.0 * ft;
+            break;
+        case 1:
+            c_impulse[N - 1] = scale * 2.0 * ft;
+            c_impulse[  N  ] = 0.0;
+            break;
+        }
+    }
+    for (i = (N + 1) / 2, j = N / 2 - 1; i < N; i++, j--)
+    {
+        posi = (double)i - m;
+        posj = (double)j - m;
+        sinc = sin (ft_rad * posi) / (PI * posi);
+        switch (wintype)
+        {
+        case 0: // Blackman-Harris 4-term
+            cosphi = cos (delta * i);
+            window  =             + 0.21747
+                    + cosphi *  ( - 0.45325
+                    + cosphi *  ( + 0.28256
+                    + cosphi *  ( - 0.04672 )));
+            break;
+        case 1: // Blackman-Harris 7-term
+            cosphi = cos (delta * i);
+            window  =             + 6.3964424114390378e-02
+                    + cosphi *  ( - 2.3993864599352804e-01
+                    + cosphi *  ( + 3.5015956323820469e-01
+                    + cosphi *  ( - 2.4774111897080783e-01
+                    + cosphi *  ( + 8.5438256055858031e-02
+                    + cosphi *  ( - 1.2320203369293225e-02
+                    + cosphi *  ( + 4.3778825791773474e-04 ))))));
+            break;
+        }
+        coef = scale * sinc * window;
+        switch (rtype)
+        {
+        case 0:
+            c_impulse[i] = + coef * cos (posi * w_osc);
+            c_impulse[j] = + coef * cos (posj * w_osc);
+            break;
+        case 1:
+            c_impulse[2 * i + 0] = + coef * cos (posi * w_osc);
+            c_impulse[2 * i + 1] = - coef * sin (posi * w_osc);
+            c_impulse[2 * j + 0] = + coef * cos (posj * w_osc);
+            c_impulse[2 * j + 1] = - coef * sin (posj * w_osc);
+            break;
+        }
+    }
+    return c_impulse;
+}
+
+double *fir_read (int N, const char *filename, int rtype, double scale)
+    // N = number of real or complex coefficients (see rtype)
+    // *filename = filename
+    // rtype = 0:  real coefficients
+    // rtype = 1:  complex coefficients
+    // scale = a scale factor that will be applied to the returned coefficients;
+    //      if this is not needed, set it to 1.0
+    // NOTE:  The number of values in the file must NOT exceed those implied by N and rtype
+{
+    FILE *file;
+    int i;
+    double I, Q;
+    double *c_impulse = (double *) malloc0 (N * sizeof (complex));
+    file = fopen (filename, "r");
+    for (i = 0; i < N; i++)
+    {
+        // read in the complex impulse response
+        // NOTE:  IF the freq response is symmetrical about 0, the imag coeffs will all be zero.
+        switch (rtype)
+        {
+        case 0:
+            fscanf (file, "%le", &I);
+            c_impulse[i] = + scale * I;
+            break;
+        case 1:
+            fscanf (file, "%le", &I);
+            fscanf (file, "%le", &Q);
+            c_impulse[2 * i + 0] = + scale * I;
+            c_impulse[2 * i + 1] = - scale * Q;
+            break;
+        }
+    }
+    fclose (file);
+    return c_impulse;
+}
+
+void analytic (int N, double* in, double* out)
+{
+    int i;
+    double inv_N = 1.0 / (double)N;
+    double two_inv_N = 2.0 * inv_N;
+    double* x = (double *) malloc0 (N * sizeof (complex));
+    fftw_plan pfor = fftw_plan_dft_1d (N, (fftw_complex *) in,
+            (fftw_complex *) x, FFTW_FORWARD, FFTW_PATIENT);
+    fftw_plan prev = fftw_plan_dft_1d (N, (fftw_complex *) x,
+            (fftw_complex *) out, FFTW_BACKWARD, FFTW_PATIENT);
+    fftw_execute (pfor);
+    x[0] *= inv_N;
+    x[1] *= inv_N;
+    for (i = 1; i < N / 2; i++)
+    {
+        x[2 * i + 0] *= two_inv_N;
+        x[2 * i + 1] *= two_inv_N;
+    }
+    x[N + 0] *= inv_N;
+    x[N + 1] *= inv_N;
+    memset (&x[N + 2], 0, (N - 2) * sizeof (double));
+    fftw_execute (prev);
+    fftw_destroy_plan (prev);
+    fftw_destroy_plan (pfor);
+    _aligned_free (x);
+}
+
+void mp_imp (int N, double* fir, double* mpfir, int pfactor, int polarity)
+{
+    int i;
+    int size = N * pfactor;
+    double inv_PN = 1.0 / (double)size;
+    double* firpad  = (double *) malloc0 (size * sizeof (complex));
+    double* firfreq = (double *) malloc0 (size * sizeof (complex));
+    double* mag     = (double *) malloc0 (size * sizeof (double));
+    double* ana     = (double *) malloc0 (size * sizeof (complex));
+    double* impulse = (double *) malloc0 (size * sizeof (complex));
+    double* newfreq = (double *) malloc0 (size * sizeof (complex));
+    memcpy (firpad, fir, N * sizeof (complex));
+    fftw_plan pfor = fftw_plan_dft_1d (size, (fftw_complex *) firpad,
+            (fftw_complex *) firfreq, FFTW_FORWARD, FFTW_PATIENT);
+    fftw_plan prev = fftw_plan_dft_1d (size, (fftw_complex *) newfreq,
+            (fftw_complex *) impulse, FFTW_BACKWARD, FFTW_PATIENT);
+    // print_impulse("orig_imp.txt", N, fir, 1, 0);
+    fftw_execute (pfor);
+    for (i = 0; i < size; i++)
+    {
+        mag[i] = sqrt (firfreq[2 * i + 0] * firfreq[2 * i + 0] + firfreq[2 * i + 1] * firfreq[2 * i + 1]) * inv_PN;
+        if (mag[i] > 0.0)
+            ana[2 * i + 0] = log (mag[i]);
+        else
+            ana[2 * i + 0] = log (1.0e-300);
+    }
+    analytic (size, ana, ana);
+    for (i = 0; i < size; i++)
+    {
+        newfreq[2 * i + 0] = + mag[i] * cos (ana[2 * i + 1]);
+        if (polarity)
+            newfreq[2 * i + 1] = + mag[i] * sin (ana[2 * i + 1]);
+        else
+            newfreq[2 * i + 1] = - mag[i] * sin (ana[2 * i + 1]);
+    }
+    fftw_execute (prev);
+    if (polarity)
+        memcpy (mpfir, &impulse[2 * (pfactor - 1) * N], N * sizeof (complex));
+    else
+        memcpy (mpfir, impulse, N * sizeof (complex));
+    // print_impulse("min_imp.txt", N, mpfir, 1, 0);
+    fftw_destroy_plan (prev);
+    fftw_destroy_plan (pfor);
+    _aligned_free (newfreq);
+    _aligned_free (impulse);
+    _aligned_free (ana);
+    _aligned_free (mag);
+    _aligned_free (firfreq);
+    _aligned_free (firpad);
+}
+
+// impulse response of a zero frequency filter comprising a cascade of two resonators,
+//    each followed by a detrending filter
+double* zff_impulse(int nc, double scale)
+{
+    // nc = number of coefficients (power of two)
+    int n_resdet = nc / 2 - 1;          // size of single zero-frequency resonator with detrender
+    int n_dresdet = 2 * n_resdet - 1;   // size of two cascaded units; when we convolve these we get 2 * n - 1 length
+    // allocate the single and make the values
+    double* resdet = (double*)malloc0 (n_resdet * sizeof(double));
+    for (int i = 1, j = 0, k = n_resdet - 1; i < nc / 4; i++, j++, k--)
+        resdet[j] = resdet[k] = (double)(i * (i + 1) / 2);
+    resdet[nc / 4 - 1] = (double)(nc / 4 * (nc / 4 + 1) / 2);
+    // print_impulse ("resdet", n_resdet, resdet, 0, 0);
+    // allocate the double and complex versions and make the values
+    double* dresdet = (double*)malloc0 (n_dresdet * sizeof(double));
+    double div = (double)((nc / 2 + 1) * (nc / 2 + 1));                 // calculate divisor
+    double* c_dresdet = (double*)malloc0 (nc * sizeof(complex));
+    for (int n = 0; n < n_dresdet; n++) // convolve to make the cascade
+    {
+        for (int k = 0; k < n_resdet; k++)
+            if ((n - k) >= 0 && (n - k) < n_resdet)
+                dresdet[n] += resdet[k] * resdet[n - k];
+        dresdet[n] /= div;
+        c_dresdet[2 * n + 0] = dresdet[n] * scale;
+        c_dresdet[2 * n + 1] = 0.0;
+    }
+    // print_impulse("dresdet", n_dresdet, dresdet, 0, 0);
+    // print_impulse("c_dresdet", nc, c_dresdet, 1, 0);
+    _aligned_free(dresdet);
+    _aligned_free(resdet);
+    return c_dresdet;
+}

wdsp/fir.h

@@ -0,0 +1,41 @@
+/*  fir.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 2016, 2022 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@pratt.one
+
+*/
+
+extern double* fftcv_mults (int NM, double* c_impulse);
+
+extern double* fir_fsamp_odd (int N, double* A, int rtype, double scale, int wintype);
+
+extern double* fir_fsamp (int N, double* A, int rtype, double scale, int wintype);
+
+extern double* fir_bandpass (int N, double f_low, double f_high, double samplerate, int wintype, int rtype, double scale);
+
+extern double* get_fsamp_window(int N, int wintype);
+
+extern double *fir_read (int N, const char *filename, int rtype, double scale);
+
+extern void mp_imp (int N, double* fir, double* mpfir, int pfactor, int polarity);
+
+extern double* zff_impulse(int nc, double scale);

wdsp/firmin.c

@@ -0,0 +1,484 @@
+/*  firmin.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 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"
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           Time-Domain FIR                                             *
+*                                                                                                       *
+********************************************************************************************************/
+
+void calc_firmin (FIRMIN a)
+{
+    a->h = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain);
+    a->rsize = a->nc;
+    a->mask = a->rsize - 1;
+    a->ring = (double *) malloc0 (a->rsize * sizeof (complex));
+    a->idx = 0;
+}
+
+FIRMIN create_firmin (int run, int position, int size, double* in, double* out,
+    int nc, double f_low, double f_high, int samplerate, int wintype, double gain)
+{
+    FIRMIN a = (FIRMIN) malloc0 (sizeof (firmin));
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->nc = nc;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    a->samplerate = samplerate;
+    a->wintype = wintype;
+    a->gain = gain;
+    calc_firmin (a);
+    return a;
+}
+
+void destroy_firmin (FIRMIN a)
+{
+    _aligned_free (a->ring);
+    _aligned_free (a->h);
+    _aligned_free (a);
+}
+
+void flush_firmin (FIRMIN a)
+{
+    memset (a->ring, 0, a->rsize * sizeof (complex));
+    a->idx = 0;
+}
+
+void xfirmin (FIRMIN a, int pos)
+{
+    if (a->run && a->position == pos)
+    {
+        int i, j, k;
+        for (i = 0; i < a->size; i++)
+        {
+            a->ring[2 * a->idx + 0] = a->in[2 * i + 0];
+            a->ring[2 * a->idx + 1] = a->in[2 * i + 1];
+            a->out[2 * i + 0] = 0.0;
+            a->out[2 * i + 1] = 0.0;
+            k = a->idx;
+            for (j = 0; j < a->nc; j++)
+            {
+                a->out[2 * i + 0] += a->h[2 * j + 0] * a->ring[2 * k + 0] - a->h[2 * j + 1] * a->ring[2 * k + 1];
+                a->out[2 * i + 1] += a->h[2 * j + 0] * a->ring[2 * k + 1] + a->h[2 * j + 1] * a->ring[2 * k + 0];
+                k = (k + a->mask) & a->mask;
+            }
+            a->idx = (a->idx + 1) & a->mask;
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_firmin (FIRMIN a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+}
+
+void setSamplerate_firmin (FIRMIN a, int rate)
+{
+    a->samplerate = (double)rate;
+    calc_firmin (a);
+}
+
+void setSize_firmin (FIRMIN a, int size)
+{
+    a->size = size;
+}
+
+void setFreqs_firmin (FIRMIN a, double f_low, double f_high)
+{
+    a->f_low = f_low;
+    a->f_high = f_high;
+    calc_firmin (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Standalone Partitioned Overlap-Save Bandpass                            *
+*                                                                                                       *
+********************************************************************************************************/
+
+void plan_firopt (FIROPT a)
+{
+    // must call for change in 'nc', 'size', 'out'
+    int i;
+    a->nfor = a->nc / a->size;
+    a->buffidx = 0;
+    a->idxmask = a->nfor - 1;
+    a->fftin = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->fftout = (double **) malloc0 (a->nfor * sizeof (double *));
+    a->fmask = (double **) malloc0 (a->nfor * sizeof (double *));
+    a->maskgen = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->pcfor = (fftw_plan *) malloc0 (a->nfor * sizeof (fftw_plan));
+    a->maskplan = (fftw_plan *) malloc0 (a->nfor * sizeof (fftw_plan));
+    for (i = 0; i < a->nfor; i++)
+    {
+        a->fftout[i] = (double *) malloc0 (2 * a->size * sizeof (complex));
+        a->fmask[i] = (double *) malloc0 (2 * a->size * sizeof (complex));
+        a->pcfor[i] = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->fftin, (fftw_complex *)a->fftout[i], FFTW_FORWARD, FFTW_PATIENT);
+        a->maskplan[i] = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->maskgen, (fftw_complex *)a->fmask[i], FFTW_FORWARD, FFTW_PATIENT);
+    }
+    a->accum = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->crev = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->accum, (fftw_complex *)a->out, FFTW_BACKWARD, FFTW_PATIENT);
+}
+
+void calc_firopt (FIROPT a)
+{
+    // call for change in frequency, rate, wintype, gain
+    // must also call after a call to plan_firopt()
+    int i;
+    double* impulse = fir_bandpass (a->nc, a->f_low, a->f_high, a->samplerate, a->wintype, 1, a->gain);
+    a->buffidx = 0;
+    for (i = 0; i < a->nfor; i++)
+    {
+        // I right-justified the impulse response => take output from left side of output buff, discard right side
+        // Be careful about flipping an asymmetrical impulse response.
+        memcpy (&(a->maskgen[2 * a->size]), &(impulse[2 * a->size * i]), a->size * sizeof(complex));
+        fftw_execute (a->maskplan[i]);
+    }
+    _aligned_free (impulse);
+}
+
+FIROPT create_firopt (int run, int position, int size, double* in, double* out,
+    int nc, double f_low, double f_high, int samplerate, int wintype, double gain)
+{
+    FIROPT a = (FIROPT) malloc0 (sizeof (firopt));
+    a->run = run;
+    a->position = position;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->nc = nc;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    a->samplerate = samplerate;
+    a->wintype = wintype;
+    a->gain = gain;
+    plan_firopt (a);
+    calc_firopt (a);
+    return a;
+}
+
+void deplan_firopt (FIROPT a)
+{
+    int i;
+    fftw_destroy_plan (a->crev);
+    _aligned_free (a->accum);
+    for (i = 0; i < a->nfor; i++)
+    {
+        _aligned_free (a->fftout[i]);
+        _aligned_free (a->fmask[i]);
+        fftw_destroy_plan (a->pcfor[i]);
+        fftw_destroy_plan (a->maskplan[i]);
+    }
+    _aligned_free (a->maskplan);
+    _aligned_free (a->pcfor);
+    _aligned_free (a->maskgen);
+    _aligned_free (a->fmask);
+    _aligned_free (a->fftout);
+    _aligned_free (a->fftin);
+}
+
+void destroy_firopt (FIROPT a)
+{
+    deplan_firopt (a);
+    _aligned_free (a);
+}
+
+void flush_firopt (FIROPT a)
+{
+    int i;
+    memset (a->fftin, 0, 2 * a->size * sizeof (complex));
+    for (i = 0; i < a->nfor; i++)
+        memset (a->fftout[i], 0, 2 * a->size * sizeof (complex));
+    a->buffidx = 0;
+}
+
+void xfiropt (FIROPT a, int pos)
+{
+    if (a->run && (a->position == pos))
+    {
+        int i, j, k;
+        memcpy (&(a->fftin[2 * a->size]), a->in, a->size * sizeof (complex));
+        fftw_execute (a->pcfor[a->buffidx]);
+        k = a->buffidx;
+        memset (a->accum, 0, 2 * a->size * sizeof (complex));
+        for (j = 0; j < a->nfor; j++)
+        {
+            for (i = 0; i < 2 * a->size; i++)
+            {
+                a->accum[2 * i + 0] += a->fftout[k][2 * i + 0] * a->fmask[j][2 * i + 0] - a->fftout[k][2 * i + 1] * a->fmask[j][2 * i + 1];
+                a->accum[2 * i + 1] += a->fftout[k][2 * i + 0] * a->fmask[j][2 * i + 1] + a->fftout[k][2 * i + 1] * a->fmask[j][2 * i + 0];
+            }
+            k = (k + a->idxmask) & a->idxmask;
+        }
+        a->buffidx = (a->buffidx + 1) & a->idxmask;
+        fftw_execute (a->crev);
+        memcpy (a->fftin, &(a->fftin[2 * a->size]), a->size * sizeof(complex));
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_firopt (FIROPT a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    deplan_firopt (a);
+    plan_firopt (a);
+    calc_firopt (a);
+}
+
+void setSamplerate_firopt (FIROPT a, int rate)
+{
+    a->samplerate = rate;
+    calc_firopt (a);
+}
+
+void setSize_firopt (FIROPT a, int size)
+{
+    a->size = size;
+    deplan_firopt (a);
+    plan_firopt (a);
+    calc_firopt (a);
+}
+
+void setFreqs_firopt (FIROPT a, double f_low, double f_high)
+{
+    a->f_low = f_low;
+    a->f_high = f_high;
+    calc_firopt (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Filter Kernel                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+
+void plan_fircore (FIRCORE a)
+{
+    // must call for change in 'nc', 'size', 'out'
+    int i;
+    a->nfor = a->nc / a->size;
+    a->cset = 0;
+    a->buffidx = 0;
+    a->idxmask = a->nfor - 1;
+    a->fftin = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->fftout   = (double **) malloc0 (a->nfor * sizeof (double *));
+    a->fmask    = (double ***) malloc0 (2 * sizeof (double **));
+    a->fmask[0] = (double **) malloc0 (a->nfor * sizeof (double *));
+    a->fmask[1] = (double **) malloc0 (a->nfor * sizeof (double *));
+    a->maskgen = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->pcfor = (fftw_plan *) malloc0 (a->nfor * sizeof (fftw_plan));
+    a->maskplan    = (fftw_plan **) malloc0 (2 * sizeof (fftw_plan *));
+    a->maskplan[0] = (fftw_plan *) malloc0 (a->nfor * sizeof (fftw_plan));
+    a->maskplan[1] = (fftw_plan *) malloc0 (a->nfor * sizeof (fftw_plan));
+    for (i = 0; i < a->nfor; i++)
+    {
+        a->fftout[i]   = (double *) malloc0 (2 * a->size * sizeof (complex));
+        a->fmask[0][i] = (double *) malloc0 (2 * a->size * sizeof (complex));
+        a->fmask[1][i] = (double *) malloc0 (2 * a->size * sizeof (complex));
+        a->pcfor[i] = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->fftin, (fftw_complex *)a->fftout[i], FFTW_FORWARD, FFTW_PATIENT);
+        a->maskplan[0][i] = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->maskgen, (fftw_complex *)a->fmask[0][i], FFTW_FORWARD, FFTW_PATIENT);
+        a->maskplan[1][i] = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->maskgen, (fftw_complex *)a->fmask[1][i], FFTW_FORWARD, FFTW_PATIENT);
+    }
+    a->accum = (double *) malloc0 (2 * a->size * sizeof (complex));
+    a->crev = fftw_plan_dft_1d(2 * a->size, (fftw_complex *)a->accum, (fftw_complex *)a->out, FFTW_BACKWARD, FFTW_PATIENT);
+    a->masks_ready = 0;
+}
+
+void calc_fircore (FIRCORE a, int flip)
+{
+    // call for change in frequency, rate, wintype, gain
+    // must also call after a call to plan_firopt()
+    int i;
+    if (a->mp)
+        mp_imp (a->nc, a->impulse, a->imp, 16, 0);
+    else
+        memcpy (a->imp, a->impulse, a->nc * sizeof (complex));
+    for (i = 0; i < a->nfor; i++)
+    {
+        // I right-justified the impulse response => take output from left side of output buff, discard right side
+        // Be careful about flipping an asymmetrical impulse response.
+        memcpy (&(a->maskgen[2 * a->size]), &(a->imp[2 * a->size * i]), a->size * sizeof(complex));
+        fftw_execute (a->maskplan[1 - a->cset][i]);
+    }
+    a->masks_ready = 1;
+    if (flip)
+    {
+        EnterCriticalSection (&a->update);
+        a->cset = 1 - a->cset;
+        LeaveCriticalSection (&a->update);
+        a->masks_ready = 0;
+    }
+}
+
+FIRCORE create_fircore (int size, double* in, double* out, int nc, int mp, double* impulse)
+{
+    FIRCORE a = (FIRCORE) malloc0 (sizeof (fircore));
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->nc = nc;
+    a->mp = mp;
+    InitializeCriticalSectionAndSpinCount (&a->update, 2500);
+    plan_fircore (a);
+    a->impulse = (double *) malloc0 (a->nc * sizeof (complex));
+    a->imp     = (double *) malloc0 (a->nc * sizeof (complex));
+    memcpy (a->impulse, impulse, a->nc * sizeof (complex));
+    calc_fircore (a, 1);
+    return a;
+}
+
+void deplan_fircore (FIRCORE a)
+{
+    int i;
+    fftw_destroy_plan (a->crev);
+    _aligned_free (a->accum);
+    for (i = 0; i < a->nfor; i++)
+    {
+        _aligned_free (a->fftout[i]);
+        _aligned_free (a->fmask[0][i]);
+        _aligned_free (a->fmask[1][i]);
+        fftw_destroy_plan (a->pcfor[i]);
+        fftw_destroy_plan (a->maskplan[0][i]);
+        fftw_destroy_plan (a->maskplan[1][i]);
+    }
+    _aligned_free (a->maskplan[0]);
+    _aligned_free (a->maskplan[1]);
+    _aligned_free (a->maskplan);
+    _aligned_free (a->pcfor);
+    _aligned_free (a->maskgen);
+    _aligned_free (a->fmask[0]);
+    _aligned_free (a->fmask[1]);
+    _aligned_free (a->fmask);
+    _aligned_free (a->fftout);
+    _aligned_free (a->fftin);
+}
+
+void destroy_fircore (FIRCORE a)
+{
+    deplan_fircore (a);
+    _aligned_free (a->imp);
+    _aligned_free (a->impulse);
+    DeleteCriticalSection (&a->update);
+    _aligned_free (a);
+}
+
+void flush_fircore (FIRCORE a)
+{
+    int i;
+    memset (a->fftin, 0, 2 * a->size * sizeof (complex));
+    for (i = 0; i < a->nfor; i++)
+        memset (a->fftout[i], 0, 2 * a->size * sizeof (complex));
+    a->buffidx = 0;
+}
+
+void xfircore (FIRCORE a)
+{
+    int i, j, k;
+    memcpy (&(a->fftin[2 * a->size]), a->in, a->size * sizeof (complex));
+    fftw_execute (a->pcfor[a->buffidx]);
+    k = a->buffidx;
+    memset (a->accum, 0, 2 * a->size * sizeof (complex));
+    EnterCriticalSection (&a->update);
+    for (j = 0; j < a->nfor; j++)
+    {
+        for (i = 0; i < 2 * a->size; i++)
+        {
+            a->accum[2 * i + 0] += a->fftout[k][2 * i + 0] * a->fmask[a->cset][j][2 * i + 0] - a->fftout[k][2 * i + 1] * a->fmask[a->cset][j][2 * i + 1];
+            a->accum[2 * i + 1] += a->fftout[k][2 * i + 0] * a->fmask[a->cset][j][2 * i + 1] + a->fftout[k][2 * i + 1] * a->fmask[a->cset][j][2 * i + 0];
+        }
+        k = (k + a->idxmask) & a->idxmask;
+    }
+    LeaveCriticalSection (&a->update);
+    a->buffidx = (a->buffidx + 1) & a->idxmask;
+    fftw_execute (a->crev);
+    memcpy (a->fftin, &(a->fftin[2 * a->size]), a->size * sizeof(complex));
+}
+
+void setBuffers_fircore (FIRCORE a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    deplan_fircore (a);
+    plan_fircore (a);
+    calc_fircore (a, 1);
+}
+
+void setSize_fircore (FIRCORE a, int size)
+{
+    a->size = size;
+    deplan_fircore (a);
+    plan_fircore (a);
+    calc_fircore (a, 1);
+}
+
+void setImpulse_fircore (FIRCORE a, double* impulse, int update)
+{
+    memcpy (a->impulse, impulse, a->nc * sizeof (complex));
+    calc_fircore (a, update);
+}
+
+void setNc_fircore (FIRCORE a, int nc, double* impulse)
+{
+    // because of FFT planning, this will probably cause a glitch in audio if done during dataflow
+    deplan_fircore (a);
+    _aligned_free (a->impulse);
+    _aligned_free (a->imp);
+    a->nc = nc;
+    plan_fircore (a);
+    a->imp     = (double *) malloc0 (a->nc * sizeof (complex));
+    a->impulse = (double *) malloc0 (a->nc * sizeof (complex));
+    memcpy (a->impulse, impulse, a->nc * sizeof (complex));
+    calc_fircore (a, 1);
+}
+
+void setMp_fircore (FIRCORE a, int mp)
+{
+    a->mp = mp;
+    calc_fircore (a, 1);
+}
+
+void setUpdate_fircore (FIRCORE a)
+{
+    if (a->masks_ready)
+    {
+        EnterCriticalSection (&a->update);
+        a->cset = 1 - a->cset;
+        LeaveCriticalSection (&a->update);
+        a->masks_ready = 0;
+    }
+}

wdsp/firmin.h

@@ -0,0 +1,184 @@
+/*  firmin.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 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
+
+*/
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           Time-Domain FIR                                             *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _firmin_h
+#define _firmin_h
+
+typedef struct _firmin
+{
+    int run;                // run control
+    int position;           // position at which to execute
+    int size;               // input/output buffer size, power of two
+    double* in;             // input buffer
+    double* out;            // output buffer, can be same as input
+    int nc;                 // number of filter coefficients, power of two
+    double f_low;           // low cutoff frequency
+    double f_high;          // high cutoff frequency
+    double* ring;           // internal complex ring buffer
+    double* h;              // complex filter coefficients
+    int rsize;              // ring size, number of complex samples, power of two
+    int mask;               // mask to update indexes
+    int idx;                // ring input/output index
+    double samplerate;      // sample rate
+    int wintype;            // filter window type
+    double gain;            // filter gain
+}firmin, *FIRMIN;
+
+extern FIRMIN create_firmin (int run, int position, int size, double* in, double* out,
+    int nc, double f_low, double f_high, int samplerate, int wintype, double gain);
+
+extern void destroy_firmin (FIRMIN a);
+
+extern void flush_firmin (FIRMIN a);
+
+extern void xfirmin (FIRMIN a, int pos);
+
+extern void setBuffers_firmin (FIRMIN a, double* in, double* out);
+
+extern void setSamplerate_firmin (FIRMIN a, int rate);
+
+extern void setSize_firmin (FIRMIN a, int size);
+
+extern void setFreqs_firmin (FIRMIN a, double f_low, double f_high);
+
+#endif
+
+/********************************************************************************************************
+*                                                                                                       *
+*                               Standalone Partitioned Overlap-Save Bandpass                            *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _firopt_h
+#define _firopt_h
+
+typedef struct _firopt
+{
+    int run;                // run control
+    int position;           // position at which to execute
+    int size;               // input/output buffer size, power of two
+    double* in;             // input buffer
+    double* out;            // output buffer, can be same as input
+    int nc;                 // number of filter coefficients, power of two, >= size
+    double f_low;           // low cutoff frequency
+    double f_high;          // high cutoff frequency
+    double samplerate;      // sample rate
+    int wintype;            // filter window type
+    double gain;            // filter gain
+    int nfor;               // number of buffers in delay line
+    double* fftin;          // fft input buffer
+    double** fmask;         // frequency domain masks
+    double** fftout;        // fftout delay line
+    double* accum;          // frequency domain accumulator
+    int buffidx;            // fft out buffer index
+    int idxmask;            // mask for index computations
+    double* maskgen;        // input for mask generation FFT
+    fftw_plan* pcfor;       // array of forward FFT plans
+    fftw_plan crev;         // reverse fft plan
+    fftw_plan* maskplan;    // plans for frequency domain masks
+} firopt, *FIROPT;
+
+extern FIROPT create_firopt (int run, int position, int size, double* in, double* out,
+    int nc, double f_low, double f_high, int samplerate, int wintype, double gain);
+
+extern void xfiropt (FIROPT a, int pos);
+
+extern void destroy_firopt (FIROPT a);
+
+extern void flush_firopt (FIROPT a);
+
+extern void setBuffers_firopt (FIROPT a, double* in, double* out);
+
+extern void setSamplerate_firopt (FIROPT a, int rate);
+
+extern void setSize_firopt (FIROPT a, int size);
+
+extern void setFreqs_firopt (FIROPT a, double f_low, double f_high);
+
+#endif
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                   Partitioned Overlap-Save Filter Kernel                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+#ifndef _fircore_h
+#define _fircore_h
+
+typedef struct _fircore
+{
+    int size;               // input/output buffer size, power of two
+    double* in;             // input buffer
+    double* out;            // output buffer, can be same as input
+    int nc;                 // number of filter coefficients, power of two, >= size
+    double* impulse;        // impulse response of filter
+    double* imp;
+    int nfor;               // number of buffers in delay line
+    double* fftin;          // fft input buffer
+    double*** fmask;        // frequency domain masks
+    double** fftout;        // fftout delay line
+    double* accum;          // frequency domain accumulator
+    int buffidx;            // fft out buffer index
+    int idxmask;            // mask for index computations
+    double* maskgen;        // input for mask generation FFT
+    fftw_plan* pcfor;       // array of forward FFT plans
+    fftw_plan crev;         // reverse fft plan
+    fftw_plan** maskplan;   // plans for frequency domain masks
+    CRITICAL_SECTION update;
+    int cset;
+    int mp;
+    int masks_ready;
+} fircore, *FIRCORE;
+
+extern FIRCORE create_fircore (int size, double* in, double* out,
+    int nc, int mp, double* impulse);
+
+extern void xfircore (FIRCORE a);
+
+extern void destroy_fircore (FIRCORE a);
+
+extern void flush_fircore (FIRCORE a);
+
+extern void setBuffers_fircore (FIRCORE a, double* in, double* out);
+
+extern void setSize_fircore (FIRCORE a, int size);
+
+extern void setImpulse_fircore (FIRCORE a, double* impulse, int update);
+
+extern void setNc_fircore (FIRCORE a, int nc, double* impulse);
+
+extern void setMp_fircore (FIRCORE a, int mp);
+
+extern void setUpdate_fircore (FIRCORE a);
+
+#endif

wdsp/fmd.c

@@ -0,0 +1,384 @@
+/*  fmd.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2013, 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"
+
+void calc_fmd (FMD a)
+{
+    // pll
+    a->omega_min = TWOPI * a->fmin / a->rate;
+    a->omega_max = TWOPI * a->fmax / a->rate;
+    a->g1 = 1.0 - exp(-2.0 * a->omegaN * a->zeta / a->rate);
+    a->g2 = -a->g1 + 2.0 * (1 - exp(-a->omegaN * a->zeta / a->rate) * cos(a->omegaN / a->rate * sqrt(1.0 - a->zeta * a->zeta)));
+    a->phs = 0.0;
+    a->fil_out = 0.0;
+    a->omega = 0.0;
+    a->pllpole = a->omegaN * sqrt(2.0 * a->zeta * a->zeta + 1.0 + sqrt((2.0 * a->zeta * a->zeta + 1.0) * (2.0 * a->zeta * a->zeta + 1.0) + 1)) / TWOPI;
+    // dc removal
+    a->mtau = exp(-1.0 / (a->rate * a->tau));
+    a->onem_mtau = 1.0 - a->mtau;
+    a->fmdc = 0.0;
+    // pll audio gain
+    a->again = a->rate / (a->deviation * TWOPI);
+    // CTCSS Removal
+    a->sntch = create_snotch(1, a->size, a->out, a->out, (int)a->rate, a->ctcss_freq, 0.0002);
+    // detector limiter
+    a->plim = create_wcpagc (
+        1,                                          // run - always ON
+        5,                                          // mode
+        1,                                          // 0 for max(I,Q), 1 for envelope
+        a->out,                                     // input buff pointer
+        a->out,                                     // output buff pointer
+        a->size,                                    // io_buffsize
+        (int)a->rate,                               // sample rate
+        0.001,                                      // tau_attack
+        0.008,                                      // tau_decay
+        4,                                          // n_tau
+        a->lim_gain,                                // max_gain (sets threshold, initial value)
+        1.0,                                        // var_gain / slope
+        1.0,                                        // fixed_gain
+        1.0,                                        // max_input
+        0.9,                                        // out_targ
+        0.250,                                      // tau_fast_backaverage
+        0.004,                                      // tau_fast_decay
+        4.0,                                        // pop_ratio
+        0,                                          // hang_enable
+        0.500,                                      // tau_hang_backmult
+        0.500,                                      // hangtime
+        2.000,                                      // hang_thresh
+        0.100);                                     // tau_hang_decay
+}
+
+void decalc_fmd (FMD a)
+{
+    destroy_wcpagc(a->plim);
+    destroy_snotch(a->sntch);
+}
+
+FMD create_fmd( int run, int size, double* in, double* out, int rate, double deviation, double f_low, double f_high,
+    double fmin, double fmax, double zeta, double omegaN, double tau, double afgain, int sntch_run, double ctcss_freq, int nc_de, int mp_de, int nc_aud, int mp_aud)
+{
+    FMD a = (FMD) malloc0 (sizeof (fmd));
+    double* impulse;
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = (double)rate;
+    a->deviation = deviation;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    a->fmin = fmin;
+    a->fmax = fmax;
+    a->zeta = zeta;
+    a->omegaN = omegaN;
+    a->tau = tau;
+    a->afgain = afgain;
+    a->sntch_run = sntch_run;
+    a->ctcss_freq = ctcss_freq;
+    a->nc_de = nc_de;
+    a->mp_de = mp_de;
+    a->nc_aud = nc_aud;
+    a->mp_aud = mp_aud;
+    a->lim_run = 0;
+    a->lim_pre_gain = 0.4;
+    a->lim_gain = 2.5;
+    calc_fmd (a);
+    // de-emphasis filter
+    a->audio = (double *) malloc0 (a->size * sizeof (complex));
+    impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    a->pde = create_fircore (a->size, a->audio, a->out, a->nc_de, a->mp_de, impulse);
+    _aligned_free (impulse);
+    // audio filter
+    impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
+    a->paud = create_fircore (a->size, a->out, a->out, a->nc_aud, a->mp_aud, impulse);
+    _aligned_free (impulse);
+    return a;
+}
+
+void destroy_fmd (FMD a)
+{
+    destroy_fircore (a->paud);
+    destroy_fircore (a->pde);
+    _aligned_free (a->audio);
+    decalc_fmd (a);
+    _aligned_free (a);
+}
+
+void flush_fmd (FMD a)
+{
+    memset (a->audio, 0, a->size * sizeof (complex));
+    flush_fircore (a->pde);
+    flush_fircore (a->paud);
+    a->phs = 0.0;
+    a->fil_out = 0.0;
+    a->omega = 0.0;
+    a->fmdc = 0.0;
+    flush_snotch (a->sntch);
+    flush_wcpagc (a->plim);
+}
+
+void xfmd (FMD a)
+{
+    if (a->run)
+    {
+        int i;
+        double det, del_out;
+        double vco[2], corr[2];
+        for (i = 0; i < a->size; i++)
+        {
+            // pll
+            vco[0]  = cos (a->phs);
+            vco[1]  = sin (a->phs);
+            corr[0] = + a->in[2 * i + 0] * vco[0] + a->in[2 * i + 1] * vco[1];
+            corr[1] = - a->in[2 * i + 0] * vco[1] + a->in[2 * i + 1] * vco[0];
+            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;
+            // dc removal, gain, & demod output
+            a->fmdc = a->mtau * a->fmdc + a->onem_mtau * a->fil_out;
+            a->audio[2 * i + 0] = a->again * (a->fil_out - a->fmdc);
+            a->audio[2 * i + 1] = a->audio[2 * i + 0];
+        }
+        // de-emphasis
+        xfircore (a->pde);
+        // audio filter
+        xfircore (a->paud);
+        // CTCSS Removal
+        xsnotch (a->sntch);
+        if (a->lim_run)
+        {
+            for (i = 0; i < 2 * a->size; i++)
+                a->out[i] *= a->lim_pre_gain;
+            xwcpagc (a->plim);
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_fmd (FMD a, double* in, double* out)
+{
+    decalc_fmd (a);
+    a->in = in;
+    a->out = out;
+    calc_fmd (a);
+    setBuffers_fircore (a->pde,  a->audio, a->out);
+    setBuffers_fircore (a->paud, a->out, a->out);
+    setBuffers_wcpagc (a->plim, a->out, a->out);
+}
+
+void setSamplerate_fmd (FMD a, int rate)
+{
+    double* impulse;
+    decalc_fmd (a);
+    a->rate = rate;
+    calc_fmd (a);
+    // de-emphasis filter
+    impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    setImpulse_fircore (a->pde, impulse, 1);
+    _aligned_free (impulse);
+    // audio filter
+    impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
+    setImpulse_fircore (a->paud, impulse, 1);
+    _aligned_free (impulse);
+    setSamplerate_wcpagc (a->plim, (int)a->rate);
+}
+
+void setSize_fmd (FMD a, int size)
+{
+    double* impulse;
+    decalc_fmd (a);
+    _aligned_free (a->audio);
+    a->size = size;
+    calc_fmd (a);
+    a->audio = (double *) malloc0 (a->size * sizeof (complex));
+    // de-emphasis filter
+    destroy_fircore (a->pde);
+    impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    a->pde = create_fircore (a->size, a->audio, a->out, a->nc_de, a->mp_de, impulse);
+    _aligned_free (impulse);
+    // audio filter
+    destroy_fircore (a->paud);
+    impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
+    a->paud = create_fircore (a->size, a->out, a->out, a->nc_aud, a->mp_aud, impulse);
+    _aligned_free (impulse);
+    setSize_wcpagc (a->plim, a->size);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXAFMDeviation (int channel, double deviation)
+{
+    FMD a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmd.p;
+    a->deviation = deviation;
+    a->again = a->rate / (a->deviation * TWOPI);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXACTCSSFreq (int channel, double freq)
+{
+    FMD a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmd.p;
+    a->ctcss_freq = freq;
+    SetSNCTCSSFreq (a->sntch, a->ctcss_freq);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXACTCSSRun (int channel, int run)
+{
+    FMD a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmd.p;
+    a->sntch_run = run;
+    SetSNCTCSSRun (a->sntch, a->sntch_run);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMNCde (int channel, int nc)
+{
+    FMD a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmd.p;
+    if (a->nc_de != nc)
+    {
+        a->nc_de = nc;
+        impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+        setNc_fircore (a->pde, a->nc_de, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMMPde (int channel, int mp)
+{
+    FMD a;
+    a = rxa[channel].fmd.p;
+    if (a->mp_de != mp)
+    {
+        a->mp_de = mp;
+        setMp_fircore (a->pde, a->mp_de);
+    }
+}
+
+PORT
+void SetRXAFMNCaud (int channel, int nc)
+{
+    FMD a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmd.p;
+    if (a->nc_aud != nc)
+    {
+        a->nc_aud = nc;
+        impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
+        setNc_fircore (a->paud, a->nc_aud, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMMPaud (int channel, int mp)
+{
+    FMD a;
+    a = rxa[channel].fmd.p;
+    if (a->mp_aud != mp)
+    {
+        a->mp_aud = mp;
+        setMp_fircore (a->paud, a->mp_aud);
+    }
+}
+
+PORT
+void SetRXAFMLimRun (int channel, int run)
+{
+    FMD a;
+    a = rxa[channel].fmd.p;
+    EnterCriticalSection(&ch[channel].csDSP);
+    if (a->lim_run != run)
+    {
+        a->lim_run = run;
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMLimGain (int channel, double gaindB)
+{
+    double gain = pow(10.0, gaindB / 20.0);
+    FMD a = rxa[channel].fmd.p;
+    EnterCriticalSection(&ch[channel].csDSP);
+    if (a->lim_gain != gain)
+    {
+        decalc_fmd(a);
+        a->lim_gain = gain;
+        calc_fmd(a);
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMAFFilter(int channel, double low, double high)
+{
+    FMD a = rxa[channel].fmd.p;
+    double* impulse;
+    EnterCriticalSection(&ch[channel].csDSP);
+    if (a->f_low != low || a->f_high != high)
+    {
+        a->f_low = low;
+        a->f_high = high;
+        // de-emphasis filter
+        impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+        setImpulse_fircore (a->pde, impulse, 1);
+        _aligned_free (impulse);
+        // audio filter
+        impulse = fir_bandpass (a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
+        setImpulse_fircore (a->paud, impulse, 1);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+}

wdsp/fmd.h

@@ -0,0 +1,110 @@
+/*  fmd.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 _fmd_h
+#define _fmd_h
+#include "iir.h"
+#include "firmin.h"
+#include "wcpAGC.h"
+typedef struct _fmd
+{
+    int run;
+    int size;
+    double* in;
+    double* out;
+    double rate;
+    double f_low;                       // audio low cutoff
+    double f_high;                      // audio high cutoff
+    // pll
+    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 pllpole;                     // pll - pole frequency
+    // for dc removal
+    double tau;
+    double mtau;
+    double onem_mtau;
+    double fmdc;
+    // pll audio gain
+    double deviation;
+    double again;
+    // for de-emphasis filter
+    double* audio;
+    FIRCORE pde;
+    int nc_de;
+    int mp_de;
+    // for audio filter
+    FIRCORE paud;
+    int nc_aud;
+    int mp_aud;
+    double afgain;
+    // CTCSS removal
+    SNOTCH sntch;
+    int sntch_run;
+    double ctcss_freq;
+    // detector limiter
+    WCPAGC plim;
+    int lim_run;
+    double lim_gain;
+    double lim_pre_gain;
+} fmd, *FMD;
+
+extern FMD create_fmd ( int run, int size, double* in, double* out, int rate, double deviation,
+    double f_low, double f_high, double fmin, double fmax, double zeta, double omegaN, double tau,
+    double afgain, int sntch_run, double ctcss_freq, int nc_de, int mp_de, int nc_aud, int mp_aud);
+
+extern void destroy_fmd (FMD a);
+
+extern void flush_fmd (FMD a);
+
+extern void xfmd (FMD a);
+
+extern void setBuffers_fmd (FMD a, double* in, double* out);
+
+extern void setSamplerate_fmd (FMD a, int rate);
+
+extern void setSize_fmd (FMD a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAFMDeviation (int channel, double deviation);
+
+extern __declspec (dllexport) void SetRXAFMNCde (int channel, int nc);
+
+extern __declspec (dllexport) void SetRXAFMMPde (int channel, int mp);
+
+extern __declspec (dllexport) void SetRXAFMNCaud (int channel, int nc);
+
+extern __declspec (dllexport) void SetRXAFMMPaud (int channel, int mp);
+
+#endif

wdsp/fmmod.c

@@ -0,0 +1,232 @@
+/*  fmmod.c
+
+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
+
+*/
+
+#include "comm.h"
+
+void calc_fmmod (FMMOD a)
+{
+    // ctcss gen
+    a->tscale = 1.0 / (1.0 + a->ctcss_level);
+    a->tphase = 0.0;
+    a->tdelta = TWOPI * a->ctcss_freq / a->samplerate;
+    // mod
+    a->sphase = 0.0;
+    a->sdelta = TWOPI * a->deviation / a->samplerate;
+    // bandpass
+    a->bp_fc = a->deviation + a->f_high;
+}
+
+FMMOD create_fmmod (int run, int size, double* in, double* out, int rate, double dev, double f_low, double f_high,
+    int ctcss_run, double ctcss_level, double ctcss_freq, int bp_run, int nc, int mp)
+{
+    FMMOD a = (FMMOD) malloc0 (sizeof (fmmod));
+    double* impulse;
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->samplerate = (double)rate;
+    a->deviation = dev;
+    a->f_low = f_low;
+    a->f_high = f_high;
+    a->ctcss_run = ctcss_run;
+    a->ctcss_level = ctcss_level;
+    a->ctcss_freq = ctcss_freq;
+    a->bp_run = bp_run;
+    a->nc = nc;
+    a->mp = mp;
+    calc_fmmod (a);
+    impulse = fir_bandpass(a->nc, -a->bp_fc, +a->bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+    a->p = create_fircore (a->size, a->out, a->out, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+    return a;
+}
+
+void destroy_fmmod (FMMOD a)
+{
+    destroy_fircore (a->p);
+    _aligned_free (a);
+}
+
+void flush_fmmod (FMMOD a)
+{
+    a->tphase = 0.0;
+    a->sphase = 0.0;
+}
+
+void xfmmod (FMMOD a)
+{
+    int i;
+    double dp, magdp, peak;
+    if (a->run)
+    {
+        peak = 0.0;
+        for (i = 0; i < a->size; i++)
+        {
+            if (a->ctcss_run)
+            {
+                a->tphase += a->tdelta;
+                if (a->tphase >= TWOPI) a->tphase -= TWOPI;
+                a->out[2 * i + 0] = a->tscale * (a->in[2 * i + 0] + a->ctcss_level * cos (a->tphase));
+            }
+            dp = a->out[2 * i + 0] * a->sdelta;
+            a->sphase += dp;
+            if (a->sphase >= TWOPI) a->sphase -= TWOPI;
+            if (a->sphase <   0.0 ) a->sphase += TWOPI;
+            a->out[2 * i + 0] = 0.7071 * cos (a->sphase);
+            a->out[2 * i + 1] = 0.7071 * sin (a->sphase);
+            if ((magdp = dp) < 0.0) magdp = - magdp;
+            if (magdp > peak) peak = magdp;
+        }
+        //print_deviation ("peakdev.txt", peak, a->samplerate);
+        if (a->bp_run)
+            xfircore (a->p);
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_fmmod (FMMOD a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+    calc_fmmod (a);
+    setBuffers_fircore (a->p, a->out, a->out);
+}
+
+void setSamplerate_fmmod (FMMOD a, int rate)
+{
+    double* impulse;
+    a->samplerate = rate;
+    calc_fmmod (a);
+    impulse = fir_bandpass(a->nc, -a->bp_fc, +a->bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+void setSize_fmmod (FMMOD a, int size)
+{
+    double* impulse;
+    a->size = size;
+    calc_fmmod (a);
+    setSize_fircore (a->p, a->size);
+    impulse = fir_bandpass(a->nc, -a->bp_fc, +a->bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+    setImpulse_fircore (a->p, impulse, 1);
+    _aligned_free (impulse);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetTXAFMDeviation (int channel, double deviation)
+{
+    FMMOD a = txa[channel].fmmod.p;
+    double bp_fc = a->f_high + deviation;
+    double* impulse = fir_bandpass (a->nc, -bp_fc, +bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+    setImpulse_fircore (a->p, impulse, 0);
+    _aligned_free (impulse);
+    EnterCriticalSection (&ch[channel].csDSP);
+    a->deviation = deviation;
+    // mod
+    a->sphase = 0.0;
+    a->sdelta = TWOPI * a->deviation / a->samplerate;
+    // bandpass
+    a->bp_fc = bp_fc;
+    setUpdate_fircore (a->p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXACTCSSFreq (int channel, double freq)
+{
+    FMMOD a;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].fmmod.p;
+    a->ctcss_freq = freq;
+    a->tphase = 0.0;
+    a->tdelta = TWOPI * a->ctcss_freq / a->samplerate;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXACTCSSRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].fmmod.p->ctcss_run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAFMNC (int channel, int nc)
+{
+    FMMOD a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = txa[channel].fmmod.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = fir_bandpass (a->nc, -a->bp_fc, +a->bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+        setNc_fircore (a->p, a->nc, impulse);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAFMMP (int channel, int mp)
+{
+    FMMOD a;
+    a = txa[channel].fmmod.p;
+    if (a->mp != mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}
+
+PORT
+void SetTXAFMAFFreqs (int channel, double low, double high)
+{
+    FMMOD a;
+    double* impulse;
+    EnterCriticalSection(&ch[channel].csDSP);
+    a = txa[channel].fmmod.p;
+    if (a->f_low != low || a->f_high != high)
+    {
+        a->f_low = low;
+        a->f_high = high;
+        a->bp_fc = a->deviation + a->f_high;
+        impulse = fir_bandpass (a->nc, -a->bp_fc, +a->bp_fc, a->samplerate, 0, 1, 1.0 / (2 * a->size));
+        setImpulse_fircore (a->p, impulse, 1);
+        _aligned_free (impulse);
+    }
+    LeaveCriticalSection(&ch[channel].csDSP);
+}

wdsp/fmmod.h

@@ -0,0 +1,87 @@
+/*  fmmod.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 _fmmod_h
+#define _fmmod_h
+#include "firmin.h"
+typedef struct _fmmod
+{
+    int run;
+    int size;
+    double* in;
+    double* out;
+    double samplerate;
+    double deviation;
+    double f_low;
+    double f_high;
+    int ctcss_run;
+    double ctcss_level;
+    double ctcss_freq;
+    // for ctcss gen
+    double tscale;
+    double tphase;
+    double tdelta;
+    // mod
+    double sphase;
+    double sdelta;
+    // bandpass
+    int bp_run;
+    double bp_fc;
+    int nc;
+    int mp;
+    FIRCORE p;
+}fmmod, *FMMOD;
+
+extern FMMOD create_fmmod (int run, int size, double* in, double* out, int rate, double dev, double f_low, double f_high,
+    int ctcss_run, double ctcss_level, double ctcss_freq, int bp_run, int nc, int mp);
+
+extern void destroy_fmmod (FMMOD a);
+
+extern void flush_fmmod (FMMOD a);
+
+extern void xfmmod (FMMOD a);
+
+extern void setBuffers_fmmod (FMMOD a, double* in, double* out);
+
+extern void setSamplerate_fmmod (FMMOD a, int rate);
+
+extern void setSize_fmmod (FMMOD a, int size);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXAFMDeviation (int channel, double deviation);
+
+extern __declspec (dllexport) void SetTXACTCSSFreq (int channel, double freq);
+
+extern __declspec (dllexport) void SetTXACTCSSRun (int channel, int run);
+
+extern __declspec (dllexport) void SetTXAFMMP (int channel, int mp);
+
+extern __declspec (dllexport) void SetTXAFMNC (int channel, int nc);
+
+extern __declspec (dllexport) void SetTXAFMAFFreqs (int channel, double low, double high);
+
+#endif

wdsp/fmsq.c

@@ -0,0 +1,279 @@
+/*  fmsq.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_fmsq (FMSQ a)
+{
+    double delta, theta;
+    double* impulse;
+    int i;
+    // noise filter
+    a->noise = (double *)malloc0(2 * a->size * sizeof(complex));
+    a->F[0] = 0.0;
+    a->F[1] = a->fc;
+    a->F[2] = *a->pllpole;
+    a->F[3] = 20000.0;
+    a->G[0] = 0.0;
+    a->G[1] = 0.0;
+    a->G[2] = 3.0;
+    a->G[3] = +20.0 * log10(20000.0 / *a->pllpole);
+    impulse = eq_impulse (a->nc, 3, a->F, a->G, a->rate, 1.0 / (2.0 * a->size), 0, 0);
+    a->p = create_fircore (a->size, a->trigger, a->noise, a->nc, a->mp, impulse);
+    _aligned_free (impulse);
+    // noise averaging
+    a->avm = exp(-1.0 / (a->rate * a->avtau));
+    a->onem_avm = 1.0 - a->avm;
+    a->avnoise = 100.0;
+    a->longavm = exp(-1.0 / (a->rate * a->longtau));
+    a->onem_longavm = 1.0 - a->longavm;
+    a->longnoise = 1.0;
+    // 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));
+    delta = PI / (double)a->ntup;
+    theta = 0.0;
+    for (i = 0; i <= a->ntup; i++)
+    {
+        a->cup[i] = 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] = 0.5 * (1 + cos(theta));
+        theta += delta;
+    }
+    // control
+    a->state = 0;
+    a->ready = 0;
+    a->ramp = 0.0;
+    a->rstep = 1.0 / a->rate;
+}
+
+void decalc_fmsq (FMSQ a)
+{
+    _aligned_free(a->cdown);
+    _aligned_free(a->cup);
+    destroy_fircore (a->p);
+    _aligned_free(a->noise);
+}
+
+FMSQ create_fmsq (int run, int size, double* insig, double* outsig, double* trigger, int rate, double fc,
+    double* pllpole, double tdelay, double avtau, double longtau, double tup, double tdown, double tail_thresh,
+    double unmute_thresh, double min_tail, double max_tail, int nc, int mp)
+{
+    FMSQ a = (FMSQ) malloc0 (sizeof (fmsq));
+    a->run = run;
+    a->size = size;
+    a->insig = insig;
+    a->outsig = outsig;
+    a->trigger = trigger;
+    a->rate = (double)rate;
+    a->fc = fc;
+    a->pllpole = pllpole;
+    a->tdelay = tdelay;
+    a->avtau = avtau;
+    a->longtau = longtau;
+    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;
+    a->nc = nc;
+    a->mp = mp;
+    calc_fmsq (a);
+    return a;
+}
+
+void destroy_fmsq (FMSQ a)
+{
+    decalc_fmsq (a);
+    _aligned_free (a);
+}
+
+void flush_fmsq (FMSQ a)
+{
+    flush_fircore (a->p);
+    a->avnoise = 100.0;
+    a->longnoise = 1.0;
+    a->state = 0;
+    a->ready = 0;
+    a->ramp = 0.0;
+}
+
+enum _fmsqstate
+{
+    MUTED,
+    INCREASE,
+    UNMUTED,
+    TAIL,
+    DECREASE
+};
+
+void xfmsq (FMSQ a)
+{
+    if (a->run)
+    {
+        int i;
+        double noise, lnlimit;
+        xfircore (a->p);
+        for (i = 0; i < a->size; i++)
+        {
+            noise = sqrt(a->noise[2 * i + 0] * a->noise[2 * i + 0] + a->noise[2 * i + 1] * a->noise[2 * i + 1]);
+            a->avnoise = a->avm * a->avnoise + a->onem_avm * noise;
+            a->longnoise = a->longavm * a->longnoise + a->onem_longavm * noise;
+            if (!a->ready) a->ramp += a->rstep;
+            if (a->ramp >= a->tdelay) a->ready = 1;
+
+            switch (a->state)
+            {
+            case MUTED:
+                if (a->avnoise < a->unmute_thresh && a->ready)
+                {
+                    a->state = INCREASE;
+                    a->count = a->ntup;
+                }
+                a->outsig[2 * i + 0] = 0.0;
+                a->outsig[2 * i + 1] = 0.0;
+                break;
+            case INCREASE:
+                a->outsig[2 * i + 0] = a->insig[2 * i + 0] * a->cup[a->ntup - a->count];
+                a->outsig[2 * i + 1] = a->insig[2 * i + 1] * a->cup[a->ntup - a->count];
+                if (a->count-- == 0)
+                    a->state = UNMUTED;
+                break;
+            case UNMUTED:
+                if (a->avnoise > a->tail_thresh)
+                {
+                    a->state = TAIL;
+                    if ((lnlimit = a->longnoise) > 1.0) lnlimit = 1.0;
+                    a->count = (int)((a->min_tail + (a->max_tail - a->min_tail) * lnlimit) * a->rate);
+                }
+                a->outsig[2 * i + 0] = a->insig[2 * i + 0];
+                a->outsig[2 * i + 1] = a->insig[2 * i + 1];
+                break;
+            case TAIL:
+                a->outsig[2 * i + 0] = a->insig[2 * i + 0];
+                a->outsig[2 * i + 1] = a->insig[2 * i + 1];
+                if (a->avnoise < a->unmute_thresh)
+                    a->state = UNMUTED;
+                else if (a->count-- == 0)
+                {
+                    a->state = DECREASE;
+                    a->count = a->ntdown;
+                }
+                break;
+            case DECREASE:
+                a->outsig[2 * i + 0] = a->insig[2 * i + 0] * a->cdown[a->ntdown - a->count];
+                a->outsig[2 * i + 1] = a->insig[2 * i + 1] * a->cdown[a->ntdown - a->count];
+                if (a->count-- == 0)
+                    a->state = MUTED;
+                break;
+            }
+        }
+    }
+    else if (a->insig != a->outsig)
+        memcpy (a->outsig, a->insig, a->size * sizeof (complex));
+}
+
+void setBuffers_fmsq (FMSQ a, double* in, double* out, double* trig)
+{
+    a->insig = in;
+    a->outsig = out;
+    a->trigger = trig;
+    setBuffers_fircore (a->p, a->trigger, a->noise);
+}
+
+void setSamplerate_fmsq (FMSQ a, int rate)
+{
+    decalc_fmsq (a);
+    a->rate = rate;
+    calc_fmsq (a);
+}
+
+void setSize_fmsq (FMSQ a, int size)
+{
+    decalc_fmsq (a);
+    a->size = size;
+    calc_fmsq (a);
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void SetRXAFMSQRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].fmsq.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMSQThreshold (int channel, double threshold)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].fmsq.p->tail_thresh = threshold;
+    rxa[channel].fmsq.p->unmute_thresh = 0.9 * threshold;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAFMSQNC (int channel, int nc)
+{
+    FMSQ a;
+    double* impulse;
+    EnterCriticalSection (&ch[channel].csDSP);
+    a = rxa[channel].fmsq.p;
+    if (a->nc != nc)
+    {
+        a->nc = nc;
+        impulse = eq_impulse (a->nc, 3, a->F, a->G, 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 SetRXAFMSQMP (int channel, int mp)
+{
+    FMSQ a;
+    a = rxa[channel].fmsq.p;
+    if (a->mp != mp)
+    {
+        a->mp = mp;
+        setMp_fircore (a->p, a->mp);
+    }
+}

wdsp/fmsq.h

@@ -0,0 +1,96 @@
+/*  fmsq.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
+
+*/
+
+#ifndef _fmsq_h
+#define _fmsq_h
+#include "firmin.h"
+typedef struct _fmsq
+{
+    int run;                            // 0 if squelch system is OFF; 1 if it's ON
+    int size;                           // size of input/output buffers
+    double* insig;                      // squelch input signal buffer
+    double* outsig;                     // squelch output signal buffer
+    double* trigger;                    // buffer used to trigger mute/unmute (may be same as input; matches timing of input buffer)
+    double rate;                        // sample rate
+    double* noise;
+    double fc;                          // corner frequency for sig / noise detection
+    double* pllpole;                    // pointer to pole frequency of the fm demodulator pll
+    double F[4];
+    double G[4];
+    double avtau;                       // time constant for averaging noise
+    double avm;
+    double onem_avm;
+    double avnoise;
+    double longtau;                     // time constant for long averaging
+    double longavm;
+    double onem_longavm;
+    double longnoise;
+    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;
+    int ready;
+    double ramp;
+    double rstep;
+    double tdelay;
+    int nc;
+    int mp;
+    FIRCORE p;
+} fmsq, *FMSQ;
+
+extern FMSQ create_fmsq (int run, int size, double* insig, double* outsig, double* trigger, int rate, double fc,
+    double* pllpole, double tdelay, double avtau, double longtau, double tup, double tdown, double tail_thresh,
+    double unmute_thresh, double min_tail, double max_tail, int nc, int mp);
+
+extern void destroy_fmsq (FMSQ a);
+
+extern void flush_fmsq (FMSQ a);
+
+extern void xfmsq (FMSQ a);
+
+extern void setBuffers_fmsq (FMSQ a, double* in, double* out, double* trig);
+
+extern void setSamplerate_fmsq (FMSQ a, int rate);
+
+extern void setSize_fmsq (FMSQ a, int size);
+
+// RXA Properties
+
+extern __declspec (dllexport) void SetRXAFMSQThreshold (int channel, double threshold);
+
+extern __declspec (dllexport) void SetRXAFMSQNC (int channel, int nc);
+
+extern __declspec (dllexport) void SetRXAFMSQMP (int channel, int mp);
+
+#endif

wdsp/gain.c

@@ -0,0 +1,125 @@
+/*  gain.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"
+
+PORT
+GAIN create_gain (int run, int* prun, int size, double* in, double* out, double Igain, double Qgain)
+{
+    GAIN a = (GAIN) malloc0 (sizeof (gain));
+    a->run = run;
+    a->prun = prun;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->Igain = Igain;
+    a->Qgain = Qgain;
+    InitializeCriticalSectionAndSpinCount(&a->cs_update, 2500);
+    return a;
+}
+
+PORT
+void destroy_gain (GAIN a)
+{
+    DeleteCriticalSection (&a->cs_update);
+    _aligned_free (a);
+}
+
+PORT
+void flush_gain (GAIN a)
+{
+
+}
+
+PORT
+void xgain (GAIN a)
+{
+    int srun;
+    EnterCriticalSection (&a->cs_update);
+    if (a->prun != 0)
+        srun = *(a->prun);
+    else
+        srun = 1;
+    if (a->run && srun)
+    {
+        int i;
+        for (i = 0; i < a->size; i++)
+        {
+            a->out[2 * i + 0] = a->Igain * a->in[2 * i + 0];
+            a->out[2 * i + 1] = a->Qgain * a->in[2 * i + 1];
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+    LeaveCriticalSection (&a->cs_update);
+}
+
+void setBuffers_gain (GAIN a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+}
+
+void setSamplerate_gain (GAIN a, int rate)
+{
+
+}
+
+void setSize_gain (GAIN a, int size)
+{
+    a->size = size;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                         POINTER-BASED PROPERTIES                                      *
+*                                                                                                       *
+********************************************************************************************************/
+
+PORT
+void pSetTXOutputLevel (GAIN a, double level)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->Igain = level;
+    a->Qgain = level;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetTXOutputLevelRun (GAIN a, int run)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->run = run;
+    LeaveCriticalSection (&a->cs_update);
+}
+
+PORT
+void pSetTXOutputLevelSize (GAIN a, int size)
+{
+    EnterCriticalSection (&a->cs_update);
+    a->size = size;
+    LeaveCriticalSection (&a->cs_update);
+}

wdsp/gain.h

@@ -0,0 +1,66 @@
+/*  gain.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 _gain_h
+#define _gain_h
+
+typedef struct _gain
+{
+    int run;
+    int* prun;
+    int size;
+    double* in;
+    double* out;
+    double Igain;
+    double Qgain;
+    CRITICAL_SECTION cs_update;
+}gain, *GAIN;
+
+__declspec (dllexport) GAIN create_gain (int run, int* prun, int size, double* in, double* out, double Igain, double Qgain);
+
+__declspec (dllexport) void destroy_gain (GAIN a);
+
+__declspec (dllexport) void flush_gain (GAIN a);
+
+__declspec (dllexport) void xgain (GAIN a);
+
+extern void setBuffers_gain (GAIN a, double* in, double* out);
+
+extern void setSamplerate_gain (GAIN a, int rate);
+
+extern void setSize_gain (GAIN a, int size);
+
+// TXA Properties
+
+// POINTER-BASED Properties
+
+__declspec (dllexport) void pSetTXOutputLevel (GAIN a, double level);
+
+__declspec (dllexport) void pSetTXOutputLevelRun (GAIN a, int run);
+
+__declspec (dllexport) void pSetTXOutputLevelSize (GAIN a, int size);
+
+#endif

wdsp/gen.c

@@ -0,0 +1,686 @@
+/*  gen.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 calc_tone (GEN a)
+{
+    a->tone.phs = 0.0;
+    a->tone.delta = TWOPI * a->tone.freq / a->rate;
+    a->tone.cosdelta = cos (a->tone.delta);
+    a->tone.sindelta = sin (a->tone.delta);
+}
+
+void calc_tt (GEN a)
+{
+    a->tt.phs1 = 0.0;
+    a->tt.phs2 = 0.0;
+    a->tt.delta1 = TWOPI * a->tt.f1 / a->rate;
+    a->tt.delta2 = TWOPI * a->tt.f2 / a->rate;
+    a->tt.cosdelta1 = cos (a->tt.delta1);
+    a->tt.cosdelta2 = cos (a->tt.delta2);
+    a->tt.sindelta1 = sin (a->tt.delta1);
+    a->tt.sindelta2 = sin (a->tt.delta2);
+}
+
+void calc_sweep (GEN a)
+{
+    a->sweep.phs = 0.0;
+    a->sweep.dphs = TWOPI * a->sweep.f1 / a->rate;
+    a->sweep.d2phs = TWOPI * a->sweep.sweeprate / (a->rate * a->rate);
+    a->sweep.dphsmax = TWOPI * a->sweep.f2 / a->rate;
+}
+
+void calc_sawtooth (GEN a)
+{
+    a->saw.period = 1.0 / a->saw.f;
+    a->saw.delta = 1.0 / a->rate;
+    a->saw.t = 0.0;
+}
+
+void calc_triangle (GEN a)
+{
+    a->tri.period = 1.0 / a->tri.f;
+    a->tri.half = 0.5 * a->tri.period;
+    a->tri.delta = 1.0 / a->rate;
+    a->tri.t = 0.0;
+    a->tri.t1 = 0.0;
+}
+
+void calc_pulse (GEN a)
+{
+    int i;
+    double delta, theta;
+    a->pulse.pperiod = 1.0 / a->pulse.pf;
+    a->pulse.tphs = 0.0;
+    a->pulse.tdelta = TWOPI * a->pulse.tf / a->rate;
+    a->pulse.tcosdelta = cos (a->pulse.tdelta);
+    a->pulse.tsindelta = sin (a->pulse.tdelta);
+    a->pulse.pntrans = (int)(a->pulse.ptranstime * a->rate);
+    a->pulse.pnon = (int)(a->pulse.pdutycycle * a->pulse.pperiod * a->rate);
+    a->pulse.pnoff = (int)(a->pulse.pperiod * a->rate) - a->pulse.pnon - 2 * a->pulse.pntrans;
+    if (a->pulse.pnoff < 0) a->pulse.pnoff = 0;
+    a->pulse.pcount = a->pulse.pnoff;
+    a->pulse.state = 0;
+    a->pulse.ctrans = (double *) malloc0 ((a->pulse.pntrans + 1) * sizeof (double));
+    delta = PI / (double)a->pulse.pntrans;
+    theta = 0.0;
+    for (i = 0; i <= a->pulse.pntrans; i++)
+    {
+        a->pulse.ctrans[i] = 0.5 * (1.0 - cos (theta));
+        theta += delta;
+    }
+}
+
+void calc_gen (GEN a)
+{
+    calc_tone (a);
+    calc_tt (a);
+    calc_sweep (a);
+    calc_sawtooth (a);
+    calc_triangle (a);
+    calc_pulse (a);
+}
+
+void decalc_gen (GEN a)
+{
+    _aligned_free (a->pulse.ctrans);
+}
+
+GEN create_gen (int run, int size, double* in, double* out, int rate, int mode)
+{
+    GEN a = (GEN) malloc0 (sizeof (gen));
+    a->run = run;
+    a->size = size;
+    a->in = in;
+    a->out = out;
+    a->rate = (double)rate;
+    a->mode = mode;
+    // tone
+    a->tone.mag = 1.0;
+    a->tone.freq = 1000.0;
+    // two-tone
+    a->tt.mag1 = 0.5;
+    a->tt.mag2 = 0.5;
+    a->tt.f1 = +  900.0;
+    a->tt.f2 = + 1700.0;
+    // noise
+    srand ((unsigned int)time (0));
+    a->noise.mag = 1.0;
+    // sweep
+    a->sweep.mag = 1.0;
+    a->sweep.f1 = -20000.0;
+    a->sweep.f2 = +20000.0;
+    a->sweep.sweeprate = +4000.0;
+    // sawtooth
+    a->saw.mag = 1.0;
+    a->saw.f = 500.0;
+    // triangle
+    a->tri.mag = 1.0;
+    a->tri.f = 500.0;
+    // pulse
+    a->pulse.mag = 1.0;
+    a->pulse.pf = 0.25;
+    a->pulse.pdutycycle = 0.25;
+    a->pulse.ptranstime = 0.002;
+    a->pulse.tf = 1000.0;
+    calc_gen (a);
+    return a;
+}
+
+void destroy_gen (GEN a)
+{
+    decalc_gen (a);
+    _aligned_free (a);
+}
+
+void flush_gen (GEN a)
+{
+    a->pulse.state = 0;
+}
+
+enum pstate
+{
+    OFF,
+    UP,
+    ON,
+    DOWN
+};
+
+void xgen (GEN a)
+{
+    if (a->run)
+    {
+        switch (a->mode)
+        {
+        case 0: // tone
+            {
+                int i;
+                double t1, t2;
+                double cosphase = cos (a->tone.phs);
+                double sinphase = sin (a->tone.phs);
+                for (i = 0; i < a->size; i++)
+                {
+                    a->out[2 * i + 0] = + a->tone.mag * cosphase;
+                    a->out[2 * i + 1] = - a->tone.mag * sinphase;
+                    t1 = cosphase;
+                    t2 = sinphase;
+                    cosphase = t1 * a->tone.cosdelta - t2 * a->tone.sindelta;
+                    sinphase = t1 * a->tone.sindelta + t2 * a->tone.cosdelta;
+                    a->tone.phs += a->tone.delta;
+                    if (a->tone.phs >= TWOPI) a->tone.phs -= TWOPI;
+                    if (a->tone.phs <   0.0 ) a->tone.phs += TWOPI;
+                }
+                break;
+            }
+        case 1: // two-tone
+            {
+                int i;
+                double tcos, tsin;
+                double cosphs1 = cos (a->tt.phs1);
+                double sinphs1 = sin (a->tt.phs1);
+                double cosphs2 = cos (a->tt.phs2);
+                double sinphs2 = sin (a->tt.phs2);
+                for (i = 0; i < a->size; i++)
+                {
+                    a->out[2 * i + 0] = + a->tt.mag1 * cosphs1 + a->tt.mag2 * cosphs2;
+                    a->out[2 * i + 1] = - a->tt.mag1 * sinphs1 - a->tt.mag2 * sinphs2;
+                    tcos = cosphs1;
+                    tsin = sinphs1;
+                    cosphs1 = tcos * a->tt.cosdelta1 - tsin * a->tt.sindelta1;
+                    sinphs1 = tcos * a->tt.sindelta1 + tsin * a->tt.cosdelta1;
+                    a->tt.phs1 += a->tt.delta1;
+                    if (a->tt.phs1 >= TWOPI) a->tt.phs1 -= TWOPI;
+                    if (a->tt.phs1 <   0.0 ) a->tt.phs1 += TWOPI;
+                    tcos = cosphs2;
+                    tsin = sinphs2;
+                    cosphs2 = tcos * a->tt.cosdelta2 - tsin * a->tt.sindelta2;
+                    sinphs2 = tcos * a->tt.sindelta2 + tsin * a->tt.cosdelta2;
+                    a->tt.phs2 += a->tt.delta2;
+                    if (a->tt.phs2 >= TWOPI) a->tt.phs2 -= TWOPI;
+                    if (a->tt.phs2 <   0.0 ) a->tt.phs2 += TWOPI;
+                }
+                break;
+            }
+        case 2: // noise
+            {
+                int i;
+                double r1, r2, c, rad;
+                for (i = 0; i < a->size; i++)
+                {
+                    do
+                    {
+                        r1 = 2.0 * (double)rand() / (double)RAND_MAX - 1.0;
+                        r2 = 2.0 * (double)rand() / (double)RAND_MAX - 1.0;
+                        c = r1 * r1 + r2 * r2;
+                    } while (c >= 1.0);
+                    rad = sqrt (-2.0 * log (c) / c);
+                    a->out[2 * i + 0] = a->noise.mag * rad * r1;
+                    a->out[2 * i + 1] = a->noise.mag * rad * r2;
+                }
+                break;
+            }
+        case 3:  // sweep
+            {
+                int i;
+                for (i = 0; i < a->size; i++)
+                {
+                    a->out[2 * i + 0] = + a->sweep.mag * cos(a->sweep.phs);
+                    a->out[2 * i + 1] = - a->sweep.mag * sin(a->sweep.phs);
+                    a->sweep.phs += a->sweep.dphs;
+                    a->sweep.dphs += a->sweep.d2phs;
+                    if (a->sweep.phs >= TWOPI) a->sweep.phs -= TWOPI;
+                    if (a->sweep.phs <   0.0 ) a->sweep.phs += TWOPI;
+                    if (a->sweep.dphs > a->sweep.dphsmax)
+                        a->sweep.dphs = TWOPI * a->sweep.f1 / a->rate;
+                }
+                break;
+            }
+        case 4:  // sawtooth (audio only)
+            {
+                int i;
+                for (i = 0; i < a->size; i++)
+                {
+                    if (a->saw.t > a->saw.period) a->saw.t -= a->saw.period;
+                    a->out[2 * i + 0] = a->saw.mag * (a->saw.t * a->saw.f - 1.0);
+                    a->out[2 * i + 1] = 0.0;
+                    a->saw.t += a->saw.delta;
+                }
+            }
+            break;
+        case 5:  // triangle (audio only)
+            {
+                int i;
+                for (i = 0; i < a->size; i++)
+                {
+                    if (a->tri.t > a->tri.period) a->tri.t1 = a->tri.t -= a->tri.period;
+                    if (a->tri.t > a->tri.half) a->tri.t1 -= a->tri.delta;
+                    else                        a->tri.t1 += a->tri.delta;
+                    a->out[2 * i + 0] = a->tri.mag * (4.0 * a->tri.t1 * a->tri.f - 1.0);
+                    a->out[2 * i + 1] = 0.0;
+                    a->tri.t += a->tri.delta;
+                }
+            }
+            break;
+        case 6:  // pulse (audio only)
+            {
+                int i;
+                double t1, t2;
+                double cosphase = cos (a->pulse.tphs);
+                double sinphase = sin (a->pulse.tphs);
+                for (i = 0; i < a->size; i++)
+                {
+                    if (a->pulse.pnoff != 0)
+                        switch (a->pulse.state)
+                        {
+                        case OFF:
+                            a->out[2 * i + 0] = 0.0;
+                            if (--a->pulse.pcount == 0)
+                            {
+                                a->pulse.state = UP;
+                                a->pulse.pcount = a->pulse.pntrans;
+                            }
+                            break;
+                        case UP:
+                            a->out[2 * i + 0] = a->pulse.mag * cosphase * a->pulse.ctrans[a->pulse.pntrans - a->pulse.pcount];
+                            if (--a->pulse.pcount == 0)
+                            {
+                                a->pulse.state = ON;
+                                a->pulse.pcount = a->pulse.pnon;
+                            }
+                            break;
+                        case ON:
+                            a->out[2 * i + 0] = a->pulse.mag * cosphase;
+                            if (--a->pulse.pcount == 0)
+                            {
+                                a->pulse.state = DOWN;
+                                a->pulse.pcount = a->pulse.pntrans;
+                            }
+                            break;
+                        case DOWN:
+                            a->out[2 * i + 0] = a->pulse.mag * cosphase * a->pulse.ctrans[a->pulse.pcount];
+                            if (--a->pulse.pcount == 0)
+                            {
+                                a->pulse.state = OFF;
+                                a->pulse.pcount = a->pulse.pnoff;
+                            }
+                            break;
+                        }
+                    else
+                        a->out[2 * i + 0] = 0.0;
+                    a->out[2 * i + 1] = 0.0;
+                    t1 = cosphase;
+                    t2 = sinphase;
+                    cosphase = t1 * a->pulse.tcosdelta - t2 * a->pulse.tsindelta;
+                    sinphase = t1 * a->pulse.tsindelta + t2 * a->pulse.tcosdelta;
+                    a->pulse.tphs += a->pulse.tdelta;
+                    if (a->pulse.tphs >= TWOPI) a->pulse.tphs -= TWOPI;
+                    if (a->pulse.tphs <   0.0 ) a->pulse.tphs += TWOPI;
+                }
+            }
+            break;
+        default:    // silence
+            {
+                memset (a->out, 0, a->size * sizeof (complex));
+                break;
+            }
+        }
+    }
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_gen (GEN a, double* in, double* out)
+{
+    a->in = in;
+    a->out = out;
+}
+
+void setSamplerate_gen (GEN a, int rate)
+{
+    decalc_gen (a);
+    a->rate = rate;
+    calc_gen (a);
+}
+
+void setSize_gen (GEN a, int size)
+{
+    a->size = size;
+    flush_gen (a);
+}
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           RXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+// 'PreGen', gen0
+
+PORT
+void SetRXAPreGenRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenMode (int channel, int mode)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->mode = mode;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenToneMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->tone.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenToneFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->tone.freq = freq;
+    calc_tone (rxa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenNoiseMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->noise.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenSweepMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->sweep.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenSweepFreq (int channel, double freq1, double freq2)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->sweep.f1 = freq1;
+    rxa[channel].gen0.p->sweep.f2 = freq2;
+    calc_sweep (rxa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetRXAPreGenSweepRate (int channel, double rate)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    rxa[channel].gen0.p->sweep.sweeprate = rate;
+    calc_sweep (rxa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+// 'PreGen', gen0
+
+PORT
+void SetTXAPreGenRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenMode (int channel, int mode)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->mode = mode;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenToneMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->tone.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenToneFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->tone.freq = freq;
+    calc_tone (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenNoiseMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->noise.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenSweepMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->sweep.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenSweepFreq (int channel, double freq1, double freq2)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->sweep.f1 = freq1;
+    txa[channel].gen0.p->sweep.f2 = freq2;
+    calc_sweep (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenSweepRate (int channel, double rate)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->sweep.sweeprate = rate;
+    calc_sweep (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenSawtoothMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->saw.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenSawtoothFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->saw.f = freq;
+    calc_sawtooth (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenTriangleMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->tri.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenTriangleFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->tri.f = freq;
+    calc_triangle (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenPulseMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->pulse.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenPulseFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->pulse.pf = freq;
+    calc_pulse (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenPulseDutyCycle (int channel, double dc)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->pulse.pdutycycle = dc;
+    calc_pulse (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenPulseToneFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->pulse.tf = freq;
+    calc_pulse (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPreGenPulseTransition (int channel, double transtime)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen0.p->pulse.ptranstime = transtime;
+    calc_pulse (txa[channel].gen0.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+// 'PostGen', gen1
+
+PORT
+void SetTXAPostGenRun (int channel, int run)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->run = run;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenMode (int channel, int mode)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->mode = mode;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenToneMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->tone.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenToneFreq (int channel, double freq)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->tone.freq = freq;
+    calc_tone (txa[channel].gen1.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenTTMag (int channel, double mag1, double mag2)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->tt.mag1 = mag1;
+    txa[channel].gen1.p->tt.mag2 = mag2;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenTTFreq (int channel, double freq1, double freq2)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->tt.f1 = freq1;
+    txa[channel].gen1.p->tt.f2 = freq2;
+    calc_tt (txa[channel].gen1.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenSweepMag (int channel, double mag)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->sweep.mag = mag;
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenSweepFreq (int channel, double freq1, double freq2)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->sweep.f1 = freq1;
+    txa[channel].gen1.p->sweep.f2 = freq2;
+    calc_sweep (txa[channel].gen1.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}
+
+PORT
+void SetTXAPostGenSweepRate (int channel, double rate)
+{
+    EnterCriticalSection (&ch[channel].csDSP);
+    txa[channel].gen1.p->sweep.sweeprate = rate;
+    calc_sweep (txa[channel].gen1.p);
+    LeaveCriticalSection (&ch[channel].csDSP);
+}

wdsp/gen.h

@@ -0,0 +1,134 @@
+/*  gen.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 _gen_h
+#define _gen_h
+
+typedef struct _gen
+{
+    int run;                    // run
+    int size;                   // number of samples per buffer
+    double* in;                 // input buffer (retained in case I want to mix in a generated signal)
+    double* out;                // output buffer
+    double rate;                // sample rate
+    int mode;
+    struct _tone
+    {
+        double mag;
+        double freq;
+        double phs;
+        double delta;
+        double cosdelta;
+        double sindelta;
+    } tone;
+    struct _tt
+    {
+        double mag1;
+        double mag2;
+        double f1;
+        double f2;
+        double phs1;
+        double phs2;
+        double delta1;
+        double delta2;
+        double cosdelta1;
+        double cosdelta2;
+        double sindelta1;
+        double sindelta2;
+    } tt;
+    struct _noise
+    {
+        double mag;
+    } noise;
+    struct _sweep
+    {
+        double mag;
+        double f1;
+        double f2;
+        double sweeprate;
+        double phs;
+        double dphs;
+        double d2phs;
+        double dphsmax;
+    } sweep;
+    struct _saw
+    {
+        double mag;
+        double f;
+        double period;
+        double delta;
+        double t;
+    } saw;
+    struct _tri
+    {
+        double mag;
+        double f;
+        double period;
+        double half;
+        double delta;
+        double t;
+        double t1;
+    } tri;
+    struct _pulse
+    {
+        double mag;
+        double pf;
+        double pdutycycle;
+        double ptranstime;
+        double* ctrans;
+        int pcount;
+        int pnon;
+        int pntrans;
+        int pnoff;
+        double pperiod;
+        double tf;
+        double tphs;
+        double tdelta;
+        double tcosdelta;
+        double tsindelta;
+        int state;
+    } pulse;
+} gen, *GEN;
+
+extern GEN create_gen (int run, int size, double* in, double* out, int rate, int mode);
+
+extern void destroy_gen (GEN a);
+
+extern void flush_gen (GEN a);
+
+extern void xgen (GEN a);
+
+extern void setBuffers_gen (GEN a, double* in, double* out);
+
+extern void setSamplerate_gen (GEN a, int rate);
+
+extern void setSize_gen (GEN a, int size);
+
+// TXA Properties
+
+extern __declspec (dllexport) void SetTXAgenRun (int channel, int run);
+
+#endif

wdsp/icfir.c

@@ -0,0 +1,221 @@
+/*  icfir.c
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2018 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@pratt.one
+
+*/
+#define _CRT_SECURE_NO_DEPRECATE
+#include "comm.h"
+
+void calc_icfir (ICFIR a)
+{
+    double* impulse;
+    a->scale = 1.0 / (double)(2 * a->size);
+    impulse = icfir_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_icfir (ICFIR a)
+{
+    destroy_fircore (a->p);
+}
+
+ICFIR create_icfir (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
+//  xbw:  width of raised cosine transition
+{
+    ICFIR a = (ICFIR) malloc0 (sizeof (icfir));
+    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_icfir (a);
+    return a;
+}
+
+void destroy_icfir (ICFIR a)
+{
+    decalc_icfir (a);
+    _aligned_free (a);
+}
+
+void flush_icfir (ICFIR a)
+{
+    flush_fircore (a->p);
+}
+
+void xicfir (ICFIR a)
+{
+    if (a->run)
+        xfircore (a->p);
+    else if (a->in != a->out)
+        memcpy (a->out, a->in, a->size * sizeof (complex));
+}
+
+void setBuffers_icfir (ICFIR a, double* in, double* out)
+{
+    decalc_icfir (a);
+    a->in = in;
+    a->out = out;
+    calc_icfir (a);
+}
+
+void setSamplerate_icfir (ICFIR a, int rate)
+{
+    decalc_icfir (a);
+    a->runrate = rate;
+    calc_icfir (a);
+}
+
+void setSize_icfir (ICFIR a, int size)
+{
+    decalc_icfir (a);
+    a->size = size;
+    calc_icfir (a);
+}
+
+void setOutRate_icfir (ICFIR a, int rate)
+{
+    decalc_icfir (a);
+    a->cicrate = rate;
+    calc_icfir (a);
+}
+
+double* icfir_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
+    // 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 = sin (PI * DD * fn) / (DD * R * sin (PI * fn / R))) < 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 = sin (PI * DD * fn) / (DD * R * sin (PI * fn / R))) < 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;
+        }
+    }
+    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 ("icfirImpulse.txt", N, impulse, 1, 0);
+    _aligned_free (A);
+    return impulse;
+}
+
+/********************************************************************************************************
+*                                                                                                       *
+*                                           TXA Properties                                              *
+*                                                                                                       *
+********************************************************************************************************/
+
+//PORT void
+//SetTXAICFIRRun (int channel, int run)
+//{
+//  EnterCriticalSection(&ch[channel].csDSP);
+//  txa[channel].icfir.p->run = run;
+//  LeaveCriticalSection(&ch[channel].csDSP);
+//}

wdsp/icfir.h

@@ -0,0 +1,71 @@
+/*  icfir.h
+
+This file is part of a program that implements a Software-Defined Radio.
+
+Copyright (C) 2018 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@pratt.one
+
+*/
+
+#ifndef _icfir_h
+#define _icfir_h
+#include "firmin.h"
+typedef struct _icfir
+{
+    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;
+} icfir, *ICFIR;
+
+extern ICFIR create_icfir (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_icfir (ICFIR a);
+
+extern void flush_icfir (ICFIR a);
+
+extern void xicfir (ICFIR a);
+
+extern void setBuffers_icfir (ICFIR a, double* in, double* out);
+
+extern void setSamplerate_icfir (ICFIR a, int rate);
+
+extern void setSize_icfir (ICFIR a, int size);
+
+extern void setOutRate_icfir (ICFIR a, int rate);
+
+extern double* icfir_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);
+
+#endif