PowerSDR Release Notes

PowerSDR/OpenHPSDR mRX PS v3.4.9 released (March 21, 2018)

(This fixes the manual frequency entry bug in v3.4.8)

This is a fairly significant release and contains many new features
including an audio adaptive variable resampler for VAC, Panafall for RX2,
and many new CAT and MIDI commands.

*General improvements:*

- Added Panafall display for RX2
- Added an Audio Adaptive Variable Resampler with monitor tools (see
below for more details)
- Corrected a resizing problem when enabling RX2
- NB/NB2 is turned OFF while transmitting when DUP is enabled
- Added 2kHz Tune Step
- Changed ANF behavior so that it is disabled when in CW mode
- Removed the 750Hz CW filter and added a 150Hz CW filter (requires
database reset to update)
- Increased display buffer to support larger than 4k displays
- Added separate VFO Lock controls for VFOA and VFOB. New VFO Lock
button will require additional skin files to operate correctly. Skins
packaged with OpenHPSDR/PowerSDR will contain the required files. You may
need to create them for other skin packages.
- Added a dropped packet ("OOOPs") counter that measures the number of
dropped receive packets from radio to PC. This may be useful in identifying
problems with network setup.
- Creates new wisdom file for each folder when using the -datapath

*MIDI interface:*

- Bug fix for Behringer mini-wheels mapping issue when mapping AGC gain
- Added support for mapping drive level to a Behringer mini-wheel
- Added support for mapping drive level to a Behringer mini-wheel.
- Added individual button mappings for VFOA and VFOB.
- Changed previous VFO Lock function to a round-robin toggle: Unlocked,
VFOA locked, VFOA&B locked, Unlocked.

*CAT interface:*

- Fixed bug in CAT Command ZZPT## to change TXProfiles in different modes
- Updated CAT Commands documentation. Found in the Documentation/Radio
- ZZUX and ZZUY locks/unlocks VFOA and VFOB, respectively. 1=lock,
- ZZVL now implements a round-robin toggle for VFO locks: Unlocked, VFOA
locked, VFOA&B locked, Unlocked.

Added the following new functions:

- ZZUS initiates a PureSignal single cal function
- ZZUT turns a two-tone test on or off (1 or 0)
- ZZGU sets RX2 AGC speed
- ZZAF,ZZAE sets VFOA N tune steps up,down respectively
- ZZBF,ZZBE sets VFOB N tune steps up,down respectively
- ZZXH sets VOX delay
- ZZCN/CO sets VFO A/B CTUN state
- ZZNU sets RX2 ANF state
- ZZXN gets combined RX1 status
- ZZXO gets combined RX2 status
- ZZXV gets combined VFO status

*Adaptive Variable Resampler:*
This release of PowerSDR mRX introduces an optional Adaptive Variable
Resampler option for VAC audio. The implementation is a true resampler and
not just a data "smoother".

Since both the radio and the PC use different clocks to obtain their
nominal 48KHz audio sampling rate, the rate in the radio will not exactly
match the rate in the PC. This sample rate mismatch leads inevitably to
audio buffer under- and over-runs that often result in audible glitches in
the VAC audio streams, both transmit and receive. The resampler acts to
transform audio data across the radio clock domain and the PC clock domain,
thereby substantially eliminating these glitches. The resampler also works
for those who are using the VAC IQ data output at all IQ sample rates, and
spur levels will be extremely low even at 192KHz.

To use the resampler, first be sure to have achieved a reasonably stable
and well performing VAC configuration without the resampler active. Then
check the "Resampler" checkbox in Setup > Audio > VAC1 (and/or VAC2 if
using VAC2). You will probably see the under- and -over-flow counters start
counting. It then takes a few seconds for the resampler algorithm to begin
making estimates of the sample rate mismatch. Once the initial estimate of
sample rate mismatch is obtained, the Var Ratio display will begin to show
the measured ratio between the PC and radio audio sample rates. This will
update in a continuous fashion and usually does not remain static, as the
sample clocks do drift over time, over temperature, etc.

At this point you should use the mouse to hover over the various displays
and controls, read the tool-tips that pop up over each one, and thereby
become more familiar with them. The "Force" controls can be left alone,
they are only there for diagnostic purposes..

After the resampler has become stable, which should occur in about ten
seconds or so, you can click on the various counters to reset them to zero.
This will allow you to more easily monitor resampler performance. Resampler
performance depends quite heavily on the performance of your particular PC
and your particular VAC configuration. Some people obtain zero under- and
over-runs in both the transmit and receive directions for many hours,
others see a steady but slow trickle that racks up to a few tens of them
per hour.

A measure of latency can also be obtained by noting the size of the
ringbuffer shown in the diagnostic display. Smaller buffer size equates to
less latency. The size of the ringbuffer is determined by an algorithm that
considers primary buffer size, VAC buffer size, VAC sample rate, and VAC
Buffer Latency settings. The smallest possible ringbuffer on the receive
side is 512, the smallest on the transmit side is 1024.

By using the monitoring features in the resampler you can work to optimize
your primary and VAC buffer settings to achieve the fewest under- and
over-runs, as well as the lowest latency. A general procedure is suggested
as follows, although this is by no means the only method. This is also a
useful procedure if you are having problems getting the resampler to
converge, i.e. you are experiencing out of control under- and over-runs.

1. Start with a large primary buffer size.
2. Start with Buffer Latency set to 0mS & "Manual".
3. Start with VAC buffer size set to match your audio interface buffer size
(if you know it, otherwise start with a large value).

If you get poor audio quality, try un-checking and checking the Buffer
Latency Manual button a few times.

4. If you absolutely can't get it to run, let Buffer Latency go back to
automatic. If that works, you can then try various values for manual buffer
latency until you find the smallest one that works for you.
5. Once things seem stable, you can experiment with reducing the primary
buffer size in order to obtain smaller ringbuffer sizes while still
maintaining good audio quality.

PowerSDR/OpenHPSDR mRX PS v3.4.2 has been released (July 5, 2017)

This release contains the following changes/fixes:

Band Stacks - import and size:
This modification adds the capability to import BandStack information from
an older database, especially useful on starting up a new version, or simply
importing while already running on a current version database. In 3.4.1
this was not yet handled. In addition, the bandstacks have been increased
to be 5-deep instead of 3.

CW Filter controls and setup:
This modification fixes the problem of CW filters not getting saved, and
worse, being lost whenever band changes or mode changes are made. It also
makes the actions of the width/shift/hi/lo/CWpitch controls all act more
consistently and intuitively, specifically for CW operation. See details
section below.

CTUN operation:
CTUN has been modified to make mode changes behave in similar ways, whether
CTUN is on or off, and are identical to the way they always worked with CTUN
off in previous versions. Behavior when tuning has also changed. As the
VFO approachs the edge of the display, instead of disappearing off the edge
or stopping, the display re-centers itself so tuning is continuous, even in
CTUN mode. The re-centering occurs as the edge of the passband hits the edge
of the display, in order to keep any signals of interest visible even as it
approaches the edge. In addition, zooming in while in CTUN mode
automatically centers the VFO in the spectrum display so that a signal of
interest (i.e. the one you're tuned to) gets zoomed in on, as is usually the
intent. When zooming out, re-centering doesn't occur, since that wouldn't
cause the VFO to disappear off the edge of the display.

CW Filter operation details:

There are several adjustments that affect the receiver filter settings in CW
modes. They are: Filter buttons, Width, Shift, High, Low, and CWPitch, and
they are somewhat interdependent.

Filter Buttons
The filter selection buttons choose pre-defined receiver bandpass filters.
They are customizable by right clicking on each button and then choosing its
width, or low and high limits. For CW it is recommended that you initially
choose a passband centered on the CW Pitch frequency, since CW filters will
automatically be centered whenever the CW Pitch is changed.

Sliding the Width control automatically switches to the Var1 filter so that
your pre-defined Filter buttons aren't changed from the width for which you
set them up and labeled them. Moving the slider left decreases the passband
width while sliding right increases it. As you increase width, one of the
passband edges (the upper edge in the "Lower"
modes such as CWL, or the lower edge in the "Upper" modes, like CWU)
approaches the limit where opposite sideband images appear (i.e. a value of
zero (0) in High or Low). When this happens, the width continues to
increase but only in one direction - downward in a "Lower" mode, and upward
in an "Upper" mode - so as to keep from hearing these images. If you
subsequently move the passband across the sideband (image) boundary,
enforcement of the boundary ceases and you can change the width centered on
wherever you've moved the passband with the Shift control. When you move the
passband back across this boundary, the width control again obeys this
limit. Clicking on a filter button other than Var1 resets the receiver to
the filter settings assigned to that button, but Var1 remains as you set it,
until you change it again, either by clicking on
Var1 or having it be automatically selected by using one of the adjustments.

Sliding the Shift control automatically switches to the Var1 filter so that
your Filter buttons aren't changed from how you set them up.
Moving the slider left shifts the passband down in frequency while moving it
right shifts the passband upward. The passband shift is not restricted the
way the Width control is and can freely slide up and down from one sideband
to the other (and affects how the Width control operates as described
above). The "Reset" button returns only the Shift slider to its original
position. Clicking on another filter button resets the receiver to the
filter settings assigned to that button, but
Var1 remains as you set it, until you change it again.

The High control shifts only the high frequency edge of the passband.
It is inactive when a pre-defined Filter button is selected, but becomes
active when Var1 or Var2 is selected. It is also possible to control it
using the CAT interface or a MIDI controller. When you do that while a
pre-set filter button is selected, Var1 is automatically selected, just as
with the Width and Shift sliders. The Low control works the same way, but
affects the low frequency edge of the passband. Mapping a MIDI controller
knob to these functions gives you a control that operates just like the Low
Cut and High Cut adjustments some transceivers provide.

CW Pitch
The CW Pitch control determines how far (in Hz) a CW signal is offset so
that it produces an audible tone when the VFO is tuned to indicate the
signal's actual frequency. Without this offset, tuning a CW signal to
zero-beat would be at the actual zero-beat point where no audible tone would
be present because its frequency is zero. Thus, in CW mode, when you tune
the VFO to a point where you hear a station's tone exactly match the CW
Pitch setting, you are tuned to transmit at that station's exact frequency.

Changing the CW Pitch control has several effects, and its interaction with
the filter buttons can get a little complicated. First, the audible tone at
"zero-beat" (i.e. when tuned so that you transmit exactly on the other
station's frequency) changes, and so does the sidetone as an aid to tuning
in a station to match the CW Pitch (and offset). Second, the CW filters are
all adjusted to keep themselves at your originally set bandwidths and
centered on the CW Pitch (offset) frequency. That way, whenever a CW signal
is tuned to its exact frequency, it's positioned in the center of the

The CW Pitch isn't usually adjusted as part of tuning in a station and
tweaking filters to reduce interference. And normally, when you customize
your CW Filter button settings, you configure all of them while keeping the
CW Pitch setting constant, using the Width (or Low and
High) setting for each button, centered around the CW Pitch frequency.
Once set that way, they will always return to these settings whenever you
choose that particular CW Pitch. When you vary the pitch from that value,
the CW filters change themselves to track the CW Pitch as described above
(but, of course, they retain their width as originally set to match their
button's label). Note, however, if you customize a CW filter button in a
way that is not centered on the CW Pitch frequency, the next time you change
CW Pitch that filter will center itself. Bandwidth takes priority over
Low/High setting values for the filter selection buttons, so that their
labels always match their bandwidths. There is one exception: If you lower
the CW Pitch below the point where the passband edge hits the sideband
(image) limit, the passband stops moving while you can continue to lower the
pitch - but the pitch/offset point will no longer be centered in the

PowerSDR/OpenHPSDR mRX PS v3.4.1 released (June 2, 2017)

Database Enhancements that you've ALL been waiting for! (Courtesy of Chris, W2PA)

To make it easier to retain settings from one release to the next, the Import Database function has been re-written. You no longer have to re-create your settings, either in the main window or in the various Setup options, when you install a new version of OpenSDR-PowerSDR mRX PS.

When you start up a new release for the first time, the program will detect that your database file was created by a previous version and attempt to create a new database version from it. The first thing that happens is a database reset, so that you start with a fresh one that's completely valid for the current release, and then the program closes as always for a reset. When you re-start it, if the old database file is intact and was working with the previous release, it should import fine and a pop-up window will tell you it succeeded. When you close this info window by clicking "OK" the program will start-up normally and have all your previous settings and options, looking just like it did before you upgraded to the new version. The old version of your database file is saved unchanged in the DB_Archive folder, and a new database.xml file replaces it in same place it was before. There should no longer be any need for Setup window screen shots.

If desired, you can still open the Setup window, manually reset the database using the "Reset Database" button, and then import another valid database file by clicking the "Import Database" button, or start personalizing from scratch as before.

The new database importer also checks for badly corrupted database files and rejects them, directing the user to reset and start over, or try importing another database file that is known to have functioned in the past. In most cases, even with a very old database file, at least some of the previous settings and options will be imported.

In addition to handling complete database files created in an old release, the new Import Database function in Setup lets you import *partial* databases, too. This means if you have a database file that contains a subset of settings or options, the Import function will simply add them to what is already present in the running database (still requiring a re-start, as usual). For example, if you have a database (XML) file containing a single Transmit Profile definition, you can send it to a friend and they can import it so they can use your customized profile too.

Therefore, to support this kind of TXProfile sharing, there is also a new button in the Setup - Transmit tab labeled "Export Current Profile." When clicked, it produces this kind of XML file containing only the currently selected Transmit Profile. The file will be stored in the same directory where your database.xml file is located, and given a file name that's the same as the exported transmit profile's name. You can then send it to another ham for import into their own database using the Import Profile button in Setup.

To summarize, the following controls in the Setup - Transmit tab support working with transmit profiles and databases:
1) The "Profiles" section at the upper left, where you can select which profile is the current one from your personal collection. This is where you can also save a profile you've just set up, under a new name, or delete one you no longer need.
2) The "More Profiles" section at the upper right. When you check the "More Profiles" checkbox, you get a scrollable list of additional default profiles that come with the installation. You can then bring the one you highlight (click on) into your own group of profiles (see #1) by clicking the "Include" button just below this list.
3) At the center extreme right side is a new button labeled "Export Current Profile" that you can use to export the currently active transmit profile (see #1) to send to someone or archive for yourself.
4) At lower left are three Database buttons that operate as before, except the "Import Database" function can now be used to import transmit profiles that were produced by the Export button - either by you or someone else. (These three buttons appear no matter what tab you've selected in setup.)

In addition to using Setup, you can select which of your saved profiles is to be used as the current (operational) profile in the main window using the pull-down list at lower right near the other audio controls, which appears when a voice mode (e.g. SSB, DSB, AM, or FM) is selected.

To export a transmit profile:
1) Click the Setup menu item to bring up the Setup window, and select the Transmit tab.
2) Select the profile you want to export using the "Profiles" pull-down list at upper left.
3) Click the "Export Current Profile" button at center, extreme right.
4) A file with the same name as the profile is written in your database directory

To import a transmit profile:
1) Click the Setup menu item to bring up the Setup window, and select the Transmit tab.
2) Click the "Import Database" button at lower left.
3) Select the database file containing the profile you want to import, and click OK
4) If the import is successful, you will be notified that the program will shut down.
5) Re-start OpenSDR-PowerSDR mRX PS.
6) The newly imported profile will appear as an additional choice in the Profiles pull-down list in Setup-Transmit.

New Transmit Modes

For some time, on the DSP => AM/SAM tab in Setup, you have been able to select to receive only one sideband when operating in SAM mode. This is useful to eliminate interference from nearby stations on one side or the other of your QSO. Now, you have a similar capability for transmit! On this same tab, you can select to transmit both sidebands (normal AM/SAM transmission) or only one of the sidebands, Upper or Lower, when operating in AM and SAM modes.

New Audio Tools

Do you have a rack of expensive audio equipment that you use to process your MIC signal before feeding it into your radio? Do you run the digital equivalent, a Digital Audio Workstation (DAW), on your computer? Well, if so, you can certainly continue to do that. However, included in this release, we are providing some important audio processing tools that enable you to IMPROVE YOUR TRANSMIT AUDIO and INCREASE THE "DENSITY" AND AVERAGE POWER of your signal without requiring these external options. The new tools are primarily a Phase Rotator and a Continuous Frequency Compressor (CFC). Accompanying these advances are additions to the TX Profiles and enhancements in the TX Equalizer and in ALC Compression control.

Rob, W1AEX, has generously prepared a "CFC Quick Setup Guide" as well as two videos, one focusing on the CFC and another focusing on the Phase Rotator. Thanks Rob!!

The Guide and Videos can be found here:

Guide: CFC Quick Setup Guide

CFC Video: <https://www.youtube.com/watch?v=j84LuuI70O4 >

Phase Rotator Video: <https://www.youtube.com/watch?v=NM2x2tk0UbY>

Also, Scott, WU2O, has developed an excellent block diagram of the TX Chain for reference during setup. It can be found here: TX Chain

  • Enhanced TX Equalizer

Equalizer frequencies are now operator-adjustable. Also note, that as it has been since the installation of the WDSP library a few years ago, this is a Continuous Gain Equalizer, NOT a multi-band equalizer. This provides smooth gain transitions across the spectrum as opposed to abrupt transitions at band edges.

  • Phase Rotator

In some AM transmitters, it is possible to boost the peak output power by having an asymmetrical audio waveform (positive peaks greater than negative peaks) and modulating to greater than 100% on positive peaks while restricting to <=100% on negative peaks to avoid "pinch-off."

In other AM transmitters and in the case of our digital-up-conversion (DUC) SDRs, this is not the correct approach. We have hard-limits which cannot be exceeded, such as the dynamic range of the DAC. The application of an asymmetrical audio waveform, with positive peaks greater than negative peaks, cannot further increase modulation in the positive direction; it instead REDUCES average power. The correct approach in such cases is to make the audio waveform as symmetrical as possible, i.e., equal positive and negative peaks.

The Phase Rotator makes the audio waveform more symmetrical. It does so by shifting the phase of various audio frequencies by varying amounts, thereby changing the shape of the waveform away from the somewhat typical asymmetrical waveform of human speech. Experimentation and analysis show that a wide range of phase shift versus frequency generally tends to improve symmetry. This wide range leads to an implementation with multiple identical stages, where the total phase shift is the sum of the shifts obtained in each stage. The stages have a specified "corner frequency" where the phase shift of the stage is equal to one-half of the total that the stage provides at the maximum frequency.

While the above explanation focused on AM, note that this feature can be used to increase average power for any speech mode. Note also, however, that it could be detrimental for any digital mode that requires coherent phase versus frequency.

Controls are found on the new DSP => CFC tab in Setup.

  • Continuous Frequency Compressor (CFC)

Many audio racks and Digital Audio Workstations provide "Multiband Compressors" which allow specifying different amounts of compression for different audio frequency bands. The CFC offers a superset of that concept where, instead of having multiple bands with constant compression in each band, the compression varies smoothly between frequency points at which it is specified. This concept is similar to the operation of the WDSP Equalizer function; however, in this case, we are varying the compression level across frequencies rather than varying the gain. A Post-CFC Equalizer is also provided as an integral part of this function to provide a final tailoring of the desired audio frequency spectrum.

Use of this function for speech modes can significantly increase the "density" and average power of the signal. Note also that use for digital modes may be detrimental, depending upon the nature of the mode.

Controls are found on the new DSP => CFC tab in Setup.

  • ALC Compression Control

On the DSP => AGC/ALC tab, you'll now find an "ALC Max Gain" control. Adjusting the gain above 0dB (the default) has a couple purposes: (1) if NOT using COMP/CESSB, it allows you to set the CFC output to peak at ~0dB and still get some compression (which is often desirable) in the ALC stage, and (2) if USING COMP/CESSB, it allows you to get some ALC compression even though the output of those stages does not exceed ~0dB.

  • CESSB Reminder

Just as a reminder, for quite some time we have had available a CESSB Overshoot Control function. This is yet another feature to increase average transmitted power at the same peak power. The CESSB algorithm was published by David Herschberger, W9GR, in the NOV/DEC 2014 issue of QEX. Dave's focus was on SSB transmission. However, in our implementation, we support its use in all voice modes. NOTE THAT A "LINEAR PHASE" TRANSMIT FILTER MUST BE SELECTED FOR THIS FUNCTION TO OPERATE PROPERLY.

This can be enabled on the Transmit tab in Setup. Note that COMP must be enabled (even at 0dB compression if you prefer) for this feature to function.

  • New TX Profiles

Need some help getting all these new audio controls set up? Some TX Profiles are provided to give you some starting points. Specifically, courtesy of Rob, W1AEX, four new TX Profiles are included with the PowerSDR mRX application. These are:
- SSB 2.8_CFC (2.8k wide - very helpful on 60 meters)
- SSB 3.0_CFC (3.0k wide)
- SSB 3.3_CFC (3.3k wide)
- AM 10_CFC (5k + 5k for 10k total width)

Other Fixes, Changes, and Enhancements

  • Four new CAT commands have been added to support the CW Audio Peaking Filter:

— ZZAP Audio Peaking Filter On/Off


— ZZAB APF Bandwidth


  • A change in the format of packets sent out by N1MM+ caused an incompatibility with the FocusMaster feature. This has been fixed.
  • The locations of XIT and RIT have been interchanged on the console.
  • There is a change in labeling of the CW Break-in checkbox. This change was made so that the label more accurately reflects the actual function of the box. This should eliminate some questions and confusion. The checkbox was previously labeled "Enabled" which would imply either ON or OFF for Break-in. Instead, what this box actually does is to allow you to select either FULL Break-in or SEMI Break-in.

PowerSDR/OpenHPSDR mRX PS v3.3.17 has been released (May 16, 2017)
- Corrects a compatibility issue with DDUtil
- Corrects the 10x watt meter reading for the Anan-10/10E transceivers

PowerSDR/OpenHPSDR mRX PS v3.3.16 has been released (May 14, 2017)

- Corrects sporadic HIGH SWR message found in v3.3.15. Your radio should
have the latest firmware flashed.
- Chris, W2PA added support for the Behringer CMD Micro and CMD PL-1 MIDI
controllers in the Midi2Cat interface. More information can be found in the
BehringerMods_Midi2Cat_v2.pdf located in the PowerSDR working directory.
- Chris, W2PA added a RIT/XIT sync feature to the console that increments
RIT and XIT the same amount when adjusting either of the values.
- Corrected an issue with the 'Limit Stitched Receivers' feature not
updating when using the 8000DLE.
- Added 200W Meter Trim range for the 8000DLE.

This release dos not implement the CFC audio tools or SSB w/carrier

PowerSDR/OpenHPSDR mRX PS v3.3.15 has been released (March 31, 2017)

- This release corrects the sporadic HIGH SWR condition present in release 3.3.14.

PowerSDR/OpenHPSDR mRX PS v3.3.14 has been released (March 26, 2017)

The major changes in this release are an update to PureSignal and the addition of the ANAN-8000DLE transceiver.

The primary objectives for PureSignal 2.0 are:

  • Make the algorithm more robust. Improve ease of automatic calibration and improve operation (1) for "difficult" amplifiers, and (2) for amplifiers being driven into heavy gain compression. ("Difficult" amplifiers are generally those with significant memory effects, typically poor plate/drain and bias voltage regulation.)
  • Streamline the user interface. In particular, (1) place Auto-Calibrate OFF/ON control and the 'Feedback' and 'Correcting' indicators on the main console, (2) reduce the screen real-estate required for the 'Linearity' window, and (3) separate 'Advanced' controls that only need to be accessed for diagnostic purposes or special situations.

You should reset your database for this new software version.


In the 'Advanced' controls section, the 'Calibration Information' panel is visible, and, there are four new controls that were not in PureSignal 1.0. These new controls represent new functionality that has been added and they should normally be left at their default settings. The controls are:

** 'PIN': Applies a priori knowledge of the required amplifier gain and phase correction in situations where the collected samples are insufficient for optimal calibration. This need can arise when the radio is not being properly driven to allow sample collection over the entire operating range, or due to some other fault or incorrect operation. This is very useful when there is a significant voltage regulation problem in an amplifier and, therefore, the gain and phase of the amplifier appear somewhat "unstable" as PureSignal collects feedback for calibration.

** 'MAP': Changes the sample-collection requirements to allow easier calibration in situations where the amplifier is detected to be in heavy gain compression. This is done by mapping the collection intervals to a different set of intervals based upon the level of compression. Note that this relaxation in sample-collection requirements MIGHT cause some degradation in high-order IMD. This is adaptive; the extent of the remapping depends upon the level of compression. If there is no compression, this function has no effect.

** 'STBL': Substantially separates the "static non-linearity" of the amplifier from the memory effects before computing the correction. This is a step toward further work on memory effects in a future release. When this is enabled, the "blue" and "yellow" displays on AmpView will show the extracted static non-linearity.

** 'TINT': A largely experimental feature allowing selection of the size of the upper interval within which samples are required to be collected. The dB value reflects the reduction from full-scale. For example, 2.5dB means ~75% of full-scale voltage or ~56% of full-scale power. This should normally be left at the default setting of 0.5dB (~94% of full-scale voltage and ~88% of full-scale power). This feature can be useful in situations where an external processing system, for example a Digital Audio Workstation, is used to precisely control the audio drive amplitude. Note, however, that Auto-Attenuate should probably be disabled when not driving to full-scale as there may not be enough information to accurately calculate its setting which can result in instability in the attenuator value.

NOTE ON AMPLIFIER GAIN COMPRESSION AND TUNING. For modes requiring linear amplification, it is NOT recommended to drive amplifiers into heavy gain compression or "flat-topping." Doing so will increase IMD significantly when PureSignal is NOT active and may increase it slightly when PureSignal is active, due to heavy reliance on the MAP function. Driving into heavy compression will not increase your peak output power by any meaningful amount and, when using PureSignal, it will not increase your average power at all since PureSignal will remove the compression in the output signal to restore linearity. For vacuum-tube amplifiers with 'Tune' and 'Load' controls, if your amplifier has excess power capability compared to your desired output level, you can generally avoid heavy gain compression by tuning the amplifier for a higher power level than your desired output and then backing down the drive level to achieve the desired output. The down-side of this tuning approach may be a slight reduction in amplifier efficiency. It should be noted that there are "proper" ways to increase your average transmit power; those include the use of the Compressor and the CESSB Overshoot Control algorithm.


PureSignal Auto-Calibrate mode can now be used without opening the "Linearity" window. Simply click the "PS-A" button in the upper-left region of the console to activate PureSignal Auto-Calibrate. The "PS-A" setting will also be remembered in the database.

Indicators for Auto-Calibrate functionality will appear immediately below the panadapter display area, on the right side. "PureSignal 2.0", when displayed in green, indicates this functionality is active. Immediately to the right of that, the "Feedback" indicator will flash each time there is a new automatic calibration and the "Correcting" indicator will light when correction is being applied. Note that the color-scheme of the PureSignal items will not follow the color scheme of the other below-panadapter information. This is because, for PureSignal, color is used to indicate the status of the various items:

OFF/ON - Blue (Auto-Calibrate OFF), Green (Auto-Calibrate ON)

Feedback Level - Red (too low), Yellow (marginally low), Green (just right), Blue (too high)

Correcting - Green (correcting), Yellow (attempting correction but with marginal feedback level), Black (no correction being applied)

Note that the "PS-A" button is dominant over the "Disable PureSignal" check-box in Setup. In other words, if you click "PS-A" to turn-on Auto-Calibrate, if "Disable PureSignal" was checked, it will become Unchecked. While "PS-A" is active, you will NOT be able to check "Disable PureSignal".

The "Linearity" window has been stream-lined. You can click "Advanced" to expand the window to show the advanced signal processing controls and indicators; however, it is expected that the typical operator will leave settings at defaults and has no need to open this section. The width of the "Linearity" window has been expanded to match the width of the "AmpView" window, making it easier to stack the two on the screen.

Minor changes in this release are:

- Bryan, W4WMT added a bug fix to the VAC feature that dramatically reduced buffer overruns when using smaller buffers.

- Corrected out of band errors for the Japan region

- Fixed CTUN so that the settings would be restored correctly after restarting the program.

- Added NR2 and SNB to the DSP menu when in the Collapsed mode.

- Added the following CAT Commands:

— ZZLI - Sets or Reads the PureSignal (PS-A) button status

— ZZNS - Sets or Reads the RX1 NR2 button status

— ZZNV - Sets or Reads the RX2 NR button status

— ZZNW - Sets or Reads the RX2 NR2 button status

PowerSDR/OpenHPSDR mRX PS v3.3.9 has been released (June 15, 2016)

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:

- waterfall display showing incorrect levels on the left hand section when
using stitched receivers. In some instances caused the program to crash.

Also includes these updates
- focus problem when setup or cwx form is open
- led font not being initialized properly
- 20dB Boost not being initialized properly

PowerSDR/OpenHPSDR mRX PS v3.3.8 has been released (June 9, 2016)

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:


Some experienced a crash in 3.3.7, especially when changing DSP Buffer Sizes
or during RX/TX transitions. We believe this has been totally resolved.


Receive latency is the time between when RF reaches your antenna and the
corresponding audio is produced in your speaker or headphones.
Similarly, transmit latency is, for example, the time between audio reaching
your microphone and RF being on its way to your antenna. For many SDRs,
especially those with sharp "brick wall" filters, the latency can be much
larger than you might expect. Depending upon the radio design and various
settings, SDR latencies can significantly exceed 100mS. Long latencies can
create problems for the operator in contest operation, high-speed break-in
CW, and even SSB rapid-turnaround VOX operation.

This release incorporates some technologies that allow us to achieve low
latencies in the same category as leading conventional radios.
Furthermore, we can do this with extremely sharp filters.

First of all, a couple basics:

- Sometime ago, we moved CW Transmit from software to the FPGA in the radio
hardware. This means that CW transmit latency was already very low, really
based upon your delay settings which are chosen to avoid any hot-switching
of relays.

- It has always been the case that the Buffer Size setting on the
Setup=>Audio/Primary tab effects latency. The lower the size, the lower the
latency. However, the lower the size, the more CPU cycles are required.
Depending upon the speed of your computer, you may be limited in how low you
can go. Fortunately, this is not likely to have such a large impact on your
latency. For a very rough estimate of the latency due to this buffer,
divide the buffer size by the sample rate. For example, a buffer of size
256, at a sample rate of 192K, contributes only about 256/192000 = 1.33mS.

As of this release, there are some new features and corresponding controls
to allow you to achieve much lower latency:

- Up until this release, "Filter Size" and "DSP Buffer Size" have been the
same and there has only been one setting, called "DSP Buffer Size."
Filter Size determines how sharp your filters are; higher filter size leads
to sharper filters. However, higher DSP Buffer Size leads to more latency
because we must collect enough samples to fill the buffer before the buffer
can be processed. As of this release, DSP Buffer Size and Filter Size are
separate and can be set by mode on the Setup=>DSP/Options tab. So, using a
very low DSP Buffer size minimizes latency and using a high Filter Size
leads to sharper filters. The trade-off here is that using lower DSP buffer
sizes requires somewhat more CPU cycles and using a high Filter Size does as
well. With a reasonably fast computer, you will likely be able to run at a
DSP Buffer Size of 64, the minimum, except, perhaps, for the FM mode. With
filter sizes of 1024 or 2048, the sharpness of our filters rival the best
radios. However, larger sizes, up to 16384, are available if you need them.

- You now have a choice of Filter Type, with two types available:
Linear Phase and Low Latency. In the past, our filters have always been
Linear Phase. Linear Phase filters have the property that all frequencies
are delayed by the same amount of time as the signal is processed through
the filter. This means that the time-domain waveform of a signal that is
totally within the passband will look the same at the input of the filter
and the output of the filter. The Low Latency filter does not strictly
comply with this same type of operation. With the Low Latency filter,
signals at frequencies very near the lower and upper edges of the passband
may experience more delay than signals at other frequencies. Comparing the
two types of filters, beta testers have reported little, if any, difference
in sound quality, no problems with several digital modes that have been
tested, and no significant negative impacts at all from using the Low
Latency filters. However, both filter types are provided for your
comparison and your choice. Of course, the Low Latency filters provide
lower latency. In fact, the latency of Linear Phase filters increases
linearly with Filter Size while the latency of the Low Latency filters is
very low and nearly independent of Filter Size.

Benchmark Comparisons:

- For CW/SSB receive, using minimum Buffer Sizes and Low Latency filters,
our beta testers have measured receive latencies in the 15mS to 20mS range.
Using minimum Buffer Sizes and Linear Phase filters, the latencies are 25mS
to 30mS for a Filter Size of 1024 and 35mS to 40mS for a Filter Size of
2048. Using features such as noise blankers, EQ, and noise reduction will
add some amount to that, depending upon the
feature(s) and settings. These numbers compare with ~65mS and ~120mS using
DSP Buffer sizes of 1024 and 2048, respectively, in prior software releases.

You WILL need to reset your database.
This release will build a new wisdom file on first time use. Depending on
your system, it may take a very long time to complete. Please be patient.

The following list of values and states where added to the TX Profiles
- selection for mic in or line in
- 20dB mic boost
- line in gain
- CESSB state
- PureSignal state

PowerSDR/OpenHPSDR mRX PS v3.3.7 has been released (April 4, 2016)

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:

- Added a completely new MIDI mapping interface from Andrew, M0YGG. This new
interface is called Midi2Cat and replaces the DJ Console midi controller
interface. It has the ability to map any midi device. We want to give Andrew
a huge thanks for sharing this very nice project.
The User Guide for MidiCat is located in the PowerSDR program folder or can
be downloaded from the Yahoo Apache Labs group files folder. Look for
'Midi2Cat_Instructions_V3.pdf'. One change to note that has changed since
the user guide was written is the Midi Interface is located on the 'CAT'
page within the Setup Form.

- Our Spectral Noise Reduction, NR2 has some significant audio improvements.
If you haven't tried it in a while, please take a moment to do so. Just as
a reminder, Noise Reduction is for Random Noise, which we all have. Also,
be sure to set your AGC Gain properly when using noise reduction —- AGC is
the "enemy" of noise reduction if the gain is set too high, i.e., if the
green "G" line is set too low.

- MNF, the Multi-Notch Filter, is now functional on transverter frequencies.

There are some enhancements and changes to the Display controls.

- First, you'll find that you can now set averaging and choice of detector
separately for the panadapter and the waterfall.

- There is now a choice of "detectors." For a digital "spectrum analyzer"
like your SDR display, a "detector" can be defined as the algorithm used to
convert FFT bins into the values displayed for the pixels across the screen.
The "Peak" detector (default) is best for signals; however, it consistently
overstates noise. If you're interested in noise measurements (band noise,
phase noise, etc.), the "Average" detector would be the optimum choice and I
recommend one of the larger FFT sizes. A "Sample" detector and "Rosenfell"
detector are also provided. To learn more about detectors, see Agilent
Application Note 150, "Spectrum Analyzer Basics," which you can find with
Google's help.
- Multiple averaging modes are provided. For general signal display and
operation, the "Log Recursive" mode (default) may be your favorite choice as
it has a visual appearance of being very responsive. For noise
measurements, you should choose one of the other modes for accuracy,
probably the "Recursive" mode is optimum. The "Log Recursive" mode will NOT
be accurate for noise measurement.
- For convenience, when either the Average or Sample detector is selected
for doing noise measurements, checking the "1 Hz BW: Av / Sa" box will
automatically normalize the displayed result to that of a 1 Hz bandwidth,
i.e., this yields the "dBm/Hz" level. This check box has no effect if
either the Peak or Rosenfell detector is selected.

Several changes to the RAForm were made to this release.
- fixed signal level discrepancies between PowerSDR S-meter readings and RA
utility readings.
- corrected the signal averaging code to take the logarithm of the averaged
power values instead of the incorrect average of the logarithms of the power
- added global region support to make file read/write functions operate
correctly in all country regions
- increased the maximum number of data points limit to 2,000,000 for each of
the time, RX1, and RX2 data arrays
- increased the maximum selectable value for x-axis display range to 100,000
- the above two new limits permit more than 27 hours of continuous data
- added control value updates on file read/write for the
"numericUpDown_mSec_between_measurements" control and the
"numericUpDown_measurements_per_point" control
- added display of Date/Time and Comment information on file READ

Other additions and fix that were added are:
- While PowerSDR is running, the Windows system timer will be set to a 1mS
precision. On exiting PowerSDR the system timer will revert back to its
previous setting.
- The 60m band frequency for the Netherlands region was changed to allow
transmit between 5.35MHz and 5.45MHz.
- Removed the DJ Console project.
- Removed the transmit low filter cutoff restriction.
- Corrected a problem when using CTUN when Stereo Diversity (SD) is enabled.
- Added a Keyboard mapping for Transmit and Receive.
- Corrected several issues with the Waterfall display.

You WILL need to reset your database.

PowerSDR/OpenHPSDR mRX PS v3.3.6 has been released (November 16, 2015)

This release can be downloaded from the openhpsdr.org website.

This release includes another tool for your Noise/Interference Toolbox.
This particular tool, Multi Notch Filter (MNF), allows you to specify up to
1024 notches. The notches are specified by RF frequency and width and will
be invoked, as needed, when they overlap the passband. This feature is
useful for those who have interference that consistently appears on specific

To avoid phase distortion, the notches are implemented with linear phase.
Also, they introduce no additional processing delay nor do they consume any
additional CPU cycles once you're on frequency and the notches are set up.
This is all accomplished by simply "cutting" the notches into the existing
bandpass filters rather than adding additional filters.

Sorry, there's currently no fancy on-panadapter UI for this. If you have
programming talents and would like to write one, please let us know.

To enter notches, in Setup, go to the new DSP / MNF tab. To Add a notch,
click Add then enter the Center Frequency and Notch Width and click Enter.
Once multiple notches are in the database, you can scroll back and forth
through them using the up/down arrows on the "Notch #"
control. When you add a new notch, it will be added immediately after the
notch being viewed. You can Edit notches by clicking Edit, making your
changes, then clicking Enter. You can Delete a notch by scrolling to the
notch and clicking Delete. If you were to enter very narrow notch widths
(less than 200Hz with default filter settings), you might not achieve the
full attenuation of >100dB. By leaving "Auto-Increase width" checked, if
your specified width is not enough to achieve >100dB notch depth, it will be
automatically increased for you when the filter is cut in the passband.
When you are entering a new notch, rather than typing in the Center
Frequency, you also have the option to tune VFO-A to the desired frequency
and then click the VFOA button.

Other changes include:
- Added CAT command for Spectral Noise Blanker ZZNN RX1 & ZZNO RX2.
- Added 25Hz Step Tune.
- Extended CAT command ZZPB to set & get 10dB, 20dB, and 30dB settings.
- Removed the subnet mask requirement for static ip addresses.
- Extended out-of-band transmit between 29.7-61.44MHz.
- Improved Russian language translation for the ToolTips by Michael, R2AGG.
- Corrected a problem with the VFO not updating when dragging the panadapter
in CW mode.

PowerSDR/OpenHPSDR mRX PS v3.2.29 has been released (October 5, 2015)

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:

- fixed open collectors not functioning during transmit
- fixed the problem with crashing using Split mode
- removed the German console language translation, ToolTip translation
- added a colon to the LED fonts

You will not need to reset the database if you have already did so for
release v3.2.28. You DO need to reset the database if you have a release
prior to v3.2.28 installed.

PowerSDR/OpenHPSDR mRX PS v3.2.28 has been released (October 4, 2015)

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:

Radio Astronomy (RA) enhancements by Joe, K5SO

Spectral Noise Blanker (SNB)

Following the release of new Noise Reduction technology, NR2, earlier this
year, this release includes another new tool for your "Noise
Toolbox": SNB, a Spectral Noise Blanker. There are many different types of
noise/interference and so having an assortment of tools at your disposal is
of great benefit. The Spectral Noise Blanker is of a very different type
than our Wideband blankers, NB and NB2, and will address some additional
impulse noise issues. It can be used along with NB/NB2 when beneficial. It
also plays quite nicely with NR/NR2 to address both Impulse Noise (SNB) and
Random Noise (NR/NR2) simultaneously.

SNB currently restricts audio bandwidth to roughly 300Hz to 5400Hz.
This was a design tradeoff which will be revisited as faster processors
become available.

To turn this blanker on, simply click the "SNB" button on the main console.
(Note that NB and NB2 have now been moved to the same
button.) In Setup, there are a couple threshold adjustments, "Threshold 1"
and "Threshold 2". However, the algorithm is self-adjusting and you
probably will not be able to achieve much improvement by fine tuning these.

Summary of your current Noise-Interference Toolbox:

Impulse Noise

  • NB (Wideband)
  • NB2 (Wideband)
  • SNB (Spectral)

Random Noise

  • NR (Variable-Leak LMS Algorithm)
  • NR2 (Signal/Noise Power Density Algorithm)

Carriers & Similar

  • ANF (Automatic Notch Filter - Variable-Leak LMS)

Noise / Interference Null

  • Diversity Reception [Requires a radio model with two ADCs]

Peaking Filters

  • Audio Peaking Filter [for CW]
  • Dolly Filter [for RTTY]

Various Bug Fixes and features
- added the ability to split the open collector outputs between VFOA (1-4)
and VFOB (5-7). To use, click on the "4x3 Split" checkbox on the
Setup=>General=>(Penny/Hermes/ANAN Ctrl) page.
- added a 'mask' field to the static IP address feature. This allows using
the correct broadcast address for the specific port the radio is on. This
corrected a problem when using a static ip address when multiple adapters
are present. See the Setup=>General=>Hardware Config page.
- added multilingual for ToolTip comments
- fixed the 6m panadapter display reading for the Anan-10E
- fixed a problem with PowerSDR crashing on startup when the Disable PA for
HF/VHF is enabled
- fixed an intermittent issue with diversity

PowerSDR/OpenHPSDR mRX PS v3.2.27 has been released (May 14, 2015)

This release can be downloaded from the apache-labs.com website.

It is not available from the openhpsdr.org website at this time.

This was released to correct a problem with the non-working CAT Commands

This does not require another database reset if you have already done so for
3.2.26. If you are upgrading from a version previous to 3.2.26 you will need
to RESET your database.

PowerSDR/OpenHPSDR mRX PS v3.2.26 has been released

This release can be downloaded from the apache-labs.com website.

It is not available from the openhpsdr.org website at this time.

This release contains the following changes:

  • You will now find the FocusMaster feature on the Setup/CAT Control/CAT

tab. This feature is aimed primarily at contest operation and automatically
restores Windows "focus" back to your logging (or other) window after you
make an adjustment in PowerSDR. There are three modes to specify a window to
which you would like focus restored: (1) if you use the N1MM+ Logger, just
select this Mode and PowerSDR will automatically find the input window, (2)
Select by Click - choose this mode and then next Click within the window to
which you'd like focus restored, and (3) Enter Window Title - type in the
text of the window titlebar EXACTLY as it appears. You can also specify a
Delay time, the time between interaction with PowerSDR and return of focus
to the selected window.

For the N1MM+ Logger mode to function, PowerSDR must receive information
from the N1MM+ Logger program identifying the input window. To enable the
N1MM+ program to send this information, a few lines must be added to the
file N1MMLogger.ini . These can be added/modified with a text editor, e.g.,
Notepad, and should read:


Windows Firewall may ask you if it's OK to allow communication between
N1MM+ and PowerSDR to which, of course, you should respond that it is.

Note also that, to allow you sufficient time to configure the FocusMaster,
focus will NOT go to your selected program as long as the Setup/CAT
Control/CAT tab is open.

  • Several operators recently reported problems using PureSignal

Auto-Calibrate with their amplifier chains. Based upon very detailed
feedback from a few people, some significant improvements have been made in
the way PureSignal deals with "challenging" amplifier chains. If you were
having difficulty with PureSignal Auto-Calibrate, we encourage you to try
this latest software release.

  • John, K5IT has shared his recent code changes that reduces latency using

the VAC feature.
Here is more information about those changes:

  • The 120ms minimum latency for VAC output has been removed.

Many users on current versions of windows can achieve much lower latency
than the default 120ms; however, if you can get frequent drop-outs setting
the value too low. 50ms is a reasonable starting point. There is a lot of
variability, so experiment to find what works best in your setup.

  • The VAC input and output audio buffers are now dynamically scaled in based

on the latency, buffer size, and sample rate.

The VAC buffer is sized to hold a maximum of twice the number of samples
required to maintain the specified average latency. For example with 48000
sample rate and 50ms latency, the buffer will hold
48000 samples/sec * 0.050 sec * 2 = 4800 samples. The buffer is then rounded
up to the next multiple of the audio buffer size. In this example if we have
selected 1024 as the audio buffer size the VAC buffer will be 4800 rounded
up to the nearest multiple of 1024: 5120 samples. The minimum VAC buffer
size is two audio buffers.

These changes are made ahead of properly implementing a variable rate
resampler which will properly and accurately match the radio's samplerate
with the audio hardware (real or virtual). The buffer scaling has been
included now after receiving favorable reports of latency improvements from
early testers.

  • Other minor bug-fixes and improvements include:

CAT Command support for ZZFR, ZZFS, and ZZCN was added.
ZZFR - Reads or sets the RX2 DSP Filter High value
ZZFS - Reads or sets the RX2 DSP Filter Low value
ZZCN - Reads or sets the CTUN button for RX1

CTUN control was added to the DJ Console Controller.

The 'BYPASS on Tx', 'Ext 2 on Tx', and 'Ext1 on Tx' work correctly when
using the "Disable BYPASS' feature.

You WILL need to reset your database.

PowerSDR/OpenHPSDR mRX PS v3.2.25 has been released.

This release contains the following changes:

  • Further Advancement for NR2. NR2 has been further enhanced with a new

"Gamma" gain method. While the change from the previous NR2 is small, this
further advance has been demonstrated to slightly further reduce background
noise and also to reduce certain audio artifacts of the noise reduction
process. After a database reset, this will become your default gain method
—- so no need to worry about selecting it.

  • PureSignal Auto-Attenuate. One of the more difficult parts of PureSignal

setup has been adjusting the attenuator during TX to optimize performance.
There is now an option (selected by default, on the Linearity window) to set
the attenuator automatically for you. When PureSignal begins a calibration,
if the attenuator is not set correctly, it will calculate a new value, turn
itself OFF, apply the new value, and turn itself back ON. In the event that
the initial condition is one of too much attenuation, the new value should
be correctly calculated and applied in a single step. In the event there is
not enough attenuation and the ADC is in overload, multiple steps may be

  • Protection for External Linear Amplifiers. Some external amplifier

manufacturers caution users that they must connect "ALC" from the amplifier
to the exciter so that the exciter will reduce drive to protect the
amplifier in the event of certain types of faults, e.g., over-current, arc,
high-SWR, etc. This has not previously been an option with openHPSDR
exciters since no connection has been provided on these exciters. Clyde,
K2UE, has led a project to develop an appropriate hardware interface for
this. Software support has been added to enable the use of this interface.
Documentation will soon be posted and the location announced. While you can
certainly build your own hardware interface from this documentation, also
stay-tuned for a possible future announcement on availability of hardware.

  • Workaround for using the Ext1/2/XVTR ports on the rev24 PA board with

receive only antennas. To use a receive only antenna with the 100W ANAN
radios shipped since Feb 2015 you will need to check the "Disable BYPASS"
box located near the bottom of the Setup=>General=>Ant/Filters=>Antenna
page. Please do not use this with previous versions of the PA boards.

  • As usual, several bug-fixes and clean-up items.


NR2 has received particularly positive feedback since its introduction a few
weeks ago. While all noise reduction algorithms tend to have some audio
artifacts, the NR2 algorithm has been reported to be quite effective in
reducing background noise without the tiring sound of typical LMS
algorithms. Several people have reported that they leave it enabled all the
time. There have also been a few questions on usage; so, we'll provide some
clarification here.

Activation: NR2 is selected using the same button as NR. From the OFF
state, click once to get NR (the normal SDR LMS type algorithm) and click
again to get NR2 (a very different type of algorithm). Clicking a third time
will get you back to the OFF state.

Noise Reduction and AGC: AGC, if not properly adjusted, is the enemy of
noise reduction algorithms! The reason is simple - AGC's job is to adjust
gain to bring things to the same level, whether those things are two
different signals or a signal and the noise during the speech pauses. So,
when NR/NR2 is trying to reduce noise, if AGC is trying to restore it, they
are in conflict. For proper operation, there are two things to observe:
(1) Set your AGC gain properly, and (2) if you experience AGC-pumping of the
noise level when using NR/NR2, try selecting the "Pre-AGC" use of NR/NR2
rather than having it "Post-AGC".
- AGC gain can most easily be correctly adjusted by displaying the green "G"
line on the panadapter and pulling it up to be a few dB above the noise
floor - experiment and you'll see how it works. To display the line, see
Setup/DSP/AGC-ALC. To move it, "grab" the green square at the left side
using your left mouse button.
- To select Pre-AGC, see Setup/DSP/NR-ANF.

Other Settings: You will note that there are other settings for NR2
available on the Setup/DSP/NR-ANF tab. I recommend you select "Gamma"
as the gain-method, either "OSMS" or "MMSE" as the NPE-method, and leave
AE-Filter checked. These settings match the default settings. In other
words, there's nothing you need to do here. On the other hand, if you wish
to experiment, and if you find some different combination that works better
in some special situation, please let us know! That will be helpful for
future developments.

You WILL need to reset your database for this release.

PowerSDR/OpenHPSDR mRX PS v3.2.24 has been released.

This release can be downloaded from the openhpsdr.org website.

This release contains the following changes:

- A new algorithm for Noise Reduction has been added. This algorithm belongs
to a different class than the other NR algorithm and we encourage you to
give it a try. In many situations, you may find it much more effective and
a more pleasant listening experience. The "NR"
buttons to enable the RX1 and RX2 Noise Reduction are of course on the main
console. To save screen real estate, these buttons are now tri-state
buttons. Click once to get the 'NR' algorithm, click again to get the new
'NR2' algorithm, and click again to turn NR2 OFF.

Tip on using Noise Reduction: Always be sure to properly set the "AGC
Gain". The simplest way to do so is to position the green "G" line just a
few dB above the noise floor on the panadapter. If you do NOT do this, the
AGC will be working to counteract the Noise Reduction. Note also a couple
facts about Noise Reduction algorithms: (1) They cannot make something out
of nothing —- they have limitations. (2) They all have some audio
"artifacts" of one kind or another.

As we continue moving toward implementation of a new software-hardware
communication protocol, we have now also made a change in the sample rates
at which the DSP internals operate. In particular, the DSP internals now
always operate at the following rates:

  • Receivers: 48K for all modes except FM, 192K for FM.
  • Transmitter: 96K for all modes.

Do not confuse these "internal DSP rates" with the selectable sample rate on
the Setup/Audio/Primary tab. You can still select 48K/96K/192K/384K as you
previously have on the Audio/Primary tab. That selection will determine the
sample rate at which the hardware returns data to the software and will
determine the panadapter width as it has in the past. So, why do the
"internal DSP rates" matter to you? They matter because the filters within
the DSP obviously operate at the internal DSP rates and their "sharpness" is
determined by the combination of sample rate and DSP Buffer Size (see
Setup/DSP/Options for buffer sizes). Recommended buffer sizes are now:

  • Receivers: >= 1024 except for, FM >= 4096.
  • Transmitter: >= 2048.

Note that before this release, if you changed the sample rate on the
Setup/Audio/Primary tab, it was also advisable to change the DSP buffer size
to compensate to provide the same filter "sharpness". That is no longer

For the NR and ANF functions, default values of "Taps" and "Delay" have also
been changed per the new DSP internal sample rates.

Note that when using PureSignal, on the Setup/Audio/Primary tab, you must
select 192K as your sample rate, just as before.

- A "NR2" control for the DJ Console was added for the new noise reduction

You WILL need to reset your database for this release.

PowerSDR/OpenHPSDR mRX PS v3.2.23 has been released.


This release contains the following bug fixes/changes:
- Corrects the sideband reversal when using the SAM sideband select feature.
- Corrects the NaN error in the SWR display.
- Corrects error in reporting negative 6m reverse power values.
- Corrects problem when tuning in the 10GHz frequency range using a DJ
- Added workaround for initializing Hermes PA relays.
- Added "Stereo Diversity" capability for Beta testing for 100D/200D radios.

PowerSDR/OpenHPSDR mRX PS v3.2.22 has been released.

This release is to fix a bug that was introduced in version 3.2.21 that
caused the program to crash when opening the XVRT Form.

A feature added to release allows the RIT to control both VFOA and VFOB when
the VFOSync is enabled.

PowerSDR/OpenHPSDR mRX PS v3.2.21

Changes to this release are:

- CESSB (Controlled Envelope Single SideBand):
Dave Hershberger, W9GR, published an interesting article on "CESSB" in the
November/December 2014 issue of QEX. The intent of his algorithm is to
increase "talk-power" by bringing the average power/speech level closer to
the peak level. This is also the algorithm that Flex chose for
incorporation in their recent 6000 series transceivers. The algorithm can
be viewed as comprising two series-connected blocks: (1) the "baseband RF
Speech Clipper," and (2) the filter overshoot control.
The first block, the "baseband RF Speech Clipper" is what we've used for
speech processor ("COMP") functionality for the last three years or so.
(Phil used this in KISS even before that.) This release adds the "filter
overshoot control" block which can be optionally enabled in Setup on the
DSP/Options tab. Note that block (2) will NOT function unless "COMP" is
enabled. However, COMP will function as before whether or not (2) is
enabled. In our implementation, this functionality is available for ALL
voice modes, not just SSB. We are grateful to several members of our group
who have tested and provided feedback on this addition! Note that not
everyone has gotten the same results by enabling item (2); this may depend
upon voice characteristics and other settings.

- RX2 DSP Buffer Size bug:
Jack, K1VT, reported what we believe to be a long-standing bug that resulted
in the DSP Buffer Size for the second receiver, RX2, to be set to the
default value rather than the database-saved value when PowerSDR was opened.
This resulted in noticeably different delays through RX1 and RX2 when
listening to the same station with both receivers. This has been fixed.

- Squelch Tail Length Control for NON-FM (SSB/AM/CW/…) modes:
This new control has been added in Setup on the DSP -> AM/SAM tab.

- In the Diversity form the 'Enable' button text was changed to 'Enabled'
with a green background when ON and 'Disabled' with a red background when

- The Bar meter (Original) can now be used in the 'Collapse' display.

- A problem with the 'Disable PA' control not taking effect immediately is

- Shortcut Keys were added that allows the CWX memories to be called without
the CWX form being open. To use, press <CNTRL> + F1-F10 in CW mode. This
change was submitted by Roberto, IK4JPN

Please reset your database.

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