latest user manual PDF

* User manual: Update reference to Experimental Features section.

* Add PR #1613 to changelog.

CMakeLists.txt

@@ -15,7 +15,7 @@ endif()
 set(CMAKE_OSX_DEPLOYMENT_TARGET "11.0" CACHE STRING "Minimum OS X deployment version")
 
 set(PROJECT_NAME FreeDV)
-set(PROJECT_VERSION 2.4.1)
+set(PROJECT_VERSION 2.4.2)
 set(PROJECT_DESCRIPTION "HF Digital Voice for Radio Amateurs")
 set(PROJECT_HOMEPAGE_URL "https://freedv.org")
 

USER_MANUAL.html

@@ -139,30 +139,30 @@ sent <strong>to</strong> your radio for transmission over the air.</p>
 <col style="width: 42%" />
 </colgroup>
 <thead>
-<tr class="header">
+<tr>
 <th>Tab</th>
 <th>Sound Device</th>
 <th>Notes</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>Receive Tab</td>
 <td>Input To Computer From Radio</td>
 <td>The off air FreeDV signal <strong>from</strong> your radio rig
 interface to your computer</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Receive Tab</td>
 <td>Output From Computer To Speaker/Headphones</td>
 <td>The decoded audio from your computer to your Speaker/Headphones</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>Transmit Tab</td>
 <td>Input From Microphone To Computer</td>
 <td>Your voice from the microphone to your computer</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Transmit Tab</td>
 <td>Output From Computer To Radio</td>
 <td>The FreeDV signal from your computer sent <strong>to</strong> your
@@ -254,21 +254,21 @@ locations depending on platform:</p>
 <col style="width: 86%" />
 </colgroup>
 <thead>
-<tr class="header">
+<tr>
 <th>Platform</th>
 <th>Typical Location</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>Windows</td>
 <td>C:\Program Files\FreeDV [version]\share\freedv-gui\wav</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Linux</td>
 <td>/usr/share/freedv-gui/wav or /usr/local/share/freedv-gui/wav</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>macOS</td>
 <td>See https://github.com/freedv/freedv-gui/tree/master/wav</td>
 </tr>
@@ -379,25 +379,25 @@ untested).</p>
 are below:</p>
 <table>
 <thead>
-<tr class="header">
+<tr>
 <th>Mode</th>
 <th>Width (kHz)</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>1600</td>
 <td>1.125</td>
 </tr>
-<tr class="even">
+<tr>
 <td>700D</td>
 <td>1.000</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>700E</td>
 <td>1.500</td>
 </tr>
-<tr class="even">
+<tr>
 <td>RADEV1</td>
 <td>1.500</td>
 </tr>
@@ -612,7 +612,7 @@ class="header-section-number">10</span> FreeDV Modes</h1>
 SSB and Skype as anchors for a rough guide to audio quality:</p>
 <table>
 <thead>
-<tr class="header">
+<tr>
 <th>Mode</th>
 <th style="text-align: center;">Min SNR</th>
 <th style="text-align: center;">Fading</th>
@@ -622,7 +622,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>SSB</td>
 <td style="text-align: center;">0</td>
 <td style="text-align: center;">8/10</td>
@@ -630,7 +630,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 <td style="text-align: center;">2600</td>
 <td style="text-align: center;">5/10</td>
 </tr>
-<tr class="even">
+<tr>
 <td>1600 (*)</td>
 <td style="text-align: center;">4</td>
 <td style="text-align: center;">3/10</td>
@@ -638,7 +638,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 <td style="text-align: center;">4000</td>
 <td style="text-align: center;">4/10</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>700D (*)</td>
 <td style="text-align: center;">-2</td>
 <td style="text-align: center;">4/10</td>
@@ -646,7 +646,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 <td style="text-align: center;">4000</td>
 <td style="text-align: center;">3/10</td>
 </tr>
-<tr class="even">
+<tr>
 <td>700E (*)</td>
 <td style="text-align: center;">1</td>
 <td style="text-align: center;">7/10</td>
@@ -654,7 +654,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 <td style="text-align: center;">4000</td>
 <td style="text-align: center;">3/10</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>RADEV1</td>
 <td style="text-align: center;">-2</td>
 <td style="text-align: center;">8/10</td>
@@ -662,7 +662,7 @@ SSB and Skype as anchors for a rough guide to audio quality:</p>
 <td style="text-align: center;">8000</td>
 <td style="text-align: center;">7/10</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Skype</td>
 <td style="text-align: center;">-</td>
 <td style="text-align: center;">-</td>
@@ -781,18 +781,18 @@ class="header-section-number">11.1</span> Tools - Filter</h2>
 <col style="width: 73%" />
 </colgroup>
 <thead>
-<tr class="header">
+<tr>
 <th>Control</th>
 <th>Description</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>Noise Suppression</td>
 <td>Enable noise suppression, de-reverberation, AGC of mic signal using
 the Speex pre-processor</td>
 </tr>
-<tr class="even">
+<tr>
 <td>700D Auto EQ</td>
 <td>Automatic equalization for FreeDV 700D/E Codec input audio</td>
 </tr>
@@ -821,18 +821,18 @@ class="header-section-number">11.2.1</span> Modem Options</h3>
 <col style="width: 71%" />
 </colgroup>
 <thead>
-<tr class="header">
+<tr>
 <th>Control</th>
 <th>Description</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>Clipping</td>
 <td>Increases the average power. Ensure your transmitter can handle high
 RMS powers before using!</td>
 </tr>
-<tr class="even">
+<tr>
 <td>TX Band Pass Filter</td>
 <td>Reduces TX spectrum bandwidth</td>
 </tr>
@@ -876,9 +876,10 @@ the menu bar by a similar process.</p>
 border and clicking and holding the left mouse button and dragging to
 required size.</p>
 <p>By default, the tab layout is not saved. However, by enabling
-experimental features (see “Advanced/Developer Features”), experimental
-support for saving and restoring the tab layout will be enabled. This
-has not been well tested and may have hidden bugs or other issues.</p>
+experimental features (see <a href="#experimental-features">Experimental
+Features</a>), experimental support for saving and restoring the tab
+layout will be enabled. This has not been well tested and may have
+hidden bugs or other issues.</p>
 <figure>
 <img src="contrib/multiple_panes.png" alt="Multiple Panes" />
 <figcaption aria-hidden="true">Multiple Panes</figcaption>
@@ -890,37 +891,37 @@ class="header-section-number">14.1</span> Stats Window</h2>
 <p>Located on the lower left hand side of the main screen.</p>
 <table>
 <thead>
-<tr class="header">
+<tr>
 <th>Term</th>
 <th>Notes</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>Bits</td>
 <td>Number of bits demodulated</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Errs</td>
 <td>Number of bit errors detected</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>Resyncs</td>
 <td>Number of times the demodulator has resynced</td>
 </tr>
-<tr class="even">
+<tr>
 <td>ClkOff</td>
 <td>Estimated sample clock offset in parts per million</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>FreqOff</td>
 <td>Estimated frequency offset in Hz</td>
 </tr>
-<tr class="even">
+<tr>
 <td>Sync</td>
 <td>Sync metric (OFDM modes like 700D/E)</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>Var</td>
 <td>Speech encoder distortion for 700D (see Auto EQ)</td>
 </tr>
@@ -1243,23 +1244,23 @@ class="header-section-number">18</span> Glossary</h1>
 <col style="width: 93%" />
 </colgroup>
 <thead>
-<tr class="header">
+<tr>
 <th>Term</th>
 <th>Notes</th>
 </tr>
 </thead>
 <tbody>
-<tr class="odd">
+<tr>
 <td>AWGN</td>
 <td>Additive White Gaussian Noise - a channel with just noise and no
 fading (like VHF)</td>
 </tr>
-<tr class="even">
+<tr>
 <td>FEC</td>
 <td>Forward Error Correction - extra bits to we send to protect the
 speech codec bits</td>
 </tr>
-<tr class="odd">
+<tr>
 <td>LDPC</td>
 <td>Low Density Parity Check Codes - a family of powerful FEC codes</td>
 </tr>
@@ -1267,8 +1268,22 @@ speech codec bits</td>
 </table>
 <h1 data-number="19" id="release-notes"><span
 class="header-section-number">19</span> Release Notes</h1>
-<h2 data-number="19.1" id="v2.4.1-tbd-2026"><span
-class="header-section-number">19.1</span> V2.4.1 TBD 2026</h2>
+<h2 data-number="19.1" id="v2.4.2-tbd-2026"><span
+class="header-section-number">19.1</span> V2.4.2 TBD 2026</h2>
+<ol type="1">
+<li>Bugfixes:
+<ul>
+<li>FreeDV Reporter: don’t select the station under the mouse when the
+list gains focus without a click. (PR #1611)</li>
+</ul></li>
+<li>Documentation:
+<ul>
+<li>User manual: Update reference to Experimental Features section. (PR
+#1613)</li>
+</ul></li>
+</ol>
+<h2 data-number="19.2" id="v2.4.1-october-2026"><span
+class="header-section-number">19.2</span> V2.4.1 October 2026</h2>
 <ol type="1">
 <li>Bugfixes:
 <ul>
@@ -1303,7 +1318,7 @@ displays. (PR #1514)</li>
 <li>Fix possible deadlock on playback completion. (PR #1533)</li>
 <li>Windows: add logic to detect audio device stalls. (PR #1537)</li>
 <li>Keep RX muted after TX until the radio confirms PTT off. (PR #1536,
-#1603)</li>
+#1603, #1605)</li>
 <li>Fix FreeDV hanging when starting the Voice Keyer with a cloud-backed
 file. (PR #1534)</li>
 <li>Fix data races on settings read by the audio threads. (PR
@@ -1424,6 +1439,39 @@ mode from starting. (PR #1594)</li>
 <li>PlotScalar: draw the trace when the plot is wider than its sample
 count. (PR #1597) - thanks <span class="citation"
 data-cites="barjac">@barjac</span>!</li>
+<li>Linux/FreeDV Reporter: fix uninitialized value deciding whether a
+selection is honored. (PR #1606)</li>
+<li>Allow microphone and speaker devices below 16 kHz (such as Bluetooth
+headsets in call mode) with a warning instead of refusing to start. (PR
+#1608)</li>
+<li>FreeDV Reporter: fix possible crash when repeatedly connecting and
+disconnecting. (freedv-backend PR #44)</li>
+<li>FreeDV Reporter: fix exiting FreeDV taking up to 60 seconds when the
+server stalls setting up the secure connection; give up on it after 10
+seconds instead. (freedv-backend PR #63)</li>
+<li>FreeDV Reporter: fix memory leak when FreeDV exits just as FreeDV
+Reporter connects. (freedv-backend PR #75)</li>
+<li>Recording: fix up to the last 100 ms of audio being lost when a
+recording stops, and log when audio has to be dropped. (freedv-backend
+PR #71, #72)</li>
+<li>Fix conversion of 16-bit audio to floating point going slightly past
+full scale. (freedv-backend PR #60)</li>
+<li>Fix buffer overruns when one RADE receive call decodes several
+frames and when resampling 8 kHz audio by large ratios. (freedv-backend
+PR #78)</li>
+<li>Fix the audio pipeline rebuilding its resampler when the sample rate
+is unchanged but not when it changes, and freeing the kept resampler
+twice. (freedv-backend PR #78)</li>
+<li>FreeDV Reporter: refuse a secure connection if TLS 1.2 or
+certificate hostname checking can’t be enforced. (freedv-backend PR
+#78)</li>
+<li>macOS/FreeDV Reporter: fix endless loop and log spam when the
+connection closes while sending. (freedv-backend PR #78)</li>
+<li>Fix network and threading code continuing after a failed socket or
+semaphore setup, Linux semaphore waits ending early on a signal, and a
+macOS crash on an empty background task. (freedv-backend PR #78)</li>
+<li>Windows: fix exiting FreeDV taking 20 seconds or more while logging
+heavily. (freedv-backend PR #78)</li>
 </ul></li>
 <li>Build system:
 <ul>
@@ -1443,6 +1491,14 @@ information, and only use ccache in CI for the builds that compile them.
 <li>Add tests of whole transmissions through the TX/RX threads without
 sound cards. (PR #1567)</li>
 <li>Resolve build failure on OpenSUSE Tumbleweed. (PR #1577)</li>
+<li>Fix Windows build failures with newer llvm-mingw (clang/libc++)
+releases. (freedv-backend PR #51)</li>
+<li>Fix Windows build with older mingw-w64 headers that don’t declare
+SetThreadDescription. (freedv-backend PR #60)</li>
+<li>macOS: fix RNNoise build failure with newer Xcode versions.
+(freedv-backend PR #59)</li>
+<li>Fix build errors with GCC 16 and with OpenSSL 4 deprecations.
+(freedv-backend PR #41, #73)</li>
 </ul></li>
 <li>Documentation:
 <ul>
@@ -1455,10 +1511,12 @@ sound cards. (PR #1567)</li>
 <li>Windows audio thread timing improvements. (PR #1488)</li>
 <li>Log when the sound server puts a PulseAudio stream on another
 device. (PR #1591)</li>
+<li>Log from the audio threads without blocking them. (freedv-backend PR
+#60)</li>
 </ul></li>
 </ol>
-<h2 data-number="19.2" id="v2.4.0-august-2026"><span
-class="header-section-number">19.2</span> V2.4.0 August 2026</h2>
+<h2 data-number="19.3" id="v2.4.0-august-2026"><span
+class="header-section-number">19.3</span> V2.4.0 August 2026</h2>
 <ol type="1">
 <li>Bugfixes:
 <ul>
@@ -1569,8 +1627,8 @@ data-cites="barjac">@barjac</span>!</li>
 #1435)</li>
 </ul></li>
 </ol>
-<h2 data-number="19.3" id="earlier-than-v2.3.0"><span
-class="header-section-number">19.3</span> Earlier than V2.3.0</h2>
+<h2 data-number="19.4" id="earlier-than-v2.3.0"><span
+class="header-section-number">19.4</span> Earlier than V2.3.0</h2>
 <p>See <a
 href="https://github.com/freedv/freedv-gui/blob/master/CHANGELOG_OLD.md">this</a>
 for more information about changes in versions prior to 2.3.0.</p>

USER_MANUAL.md

@@ -618,7 +618,7 @@ As the Tab is dragged into position a faint blue/gray image will show the positi
 
 Tabs can be resized as required by hovering the cursor over the border and clicking and holding the left mouse button and dragging to required size.
 
-By default, the tab layout is not saved. However, by enabling experimental features (see "Advanced/Developer Features"), experimental support for
+By default, the tab layout is not saved. However, by enabling experimental features (see [Experimental Features](#experimental-features)), experimental support for
 saving and restoring the tab layout will be enabled. This has not been well tested and may have hidden bugs or other issues.
 
 ![Multiple Panes](contrib/multiple_panes.png)
@@ -947,7 +947,14 @@ LDPC | Low Density Parity Check Codes - a family of powerful FEC codes
 
 # Release Notes
 
-## V2.4.1 TBD 2026
+## V2.4.2 TBD 2026
+
+1. Bugfixes:
+    * FreeDV Reporter: don't select the station under the mouse when the list gains focus without a click. (PR #1611)
+2. Documentation:
+    * User manual: Update reference to Experimental Features section. (PR #1613)
+
+## V2.4.1 October 2026
 
 1. Bugfixes:
     * Add logic to prevent FIFO sizes that are too small to allow TX thread to function. (PR #1474)
@@ -1026,6 +1033,19 @@ LDPC | Low Density Parity Check Codes - a family of powerful FEC codes
     * Stop the radio recording budget at zero instead of wrapping it. (PR #1592)
     * Windows: Fix issue preventing audio devices that don't support raw mode from starting. (PR #1594)
     * PlotScalar: draw the trace when the plot is wider than its sample count. (PR #1597) - thanks @barjac!
+    * Linux/FreeDV Reporter: fix uninitialized value deciding whether a selection is honored. (PR #1606)
+    * Allow microphone and speaker devices below 16 kHz (such as Bluetooth headsets in call mode) with a warning instead of refusing to start. (PR #1608)
+    * FreeDV Reporter: fix possible crash when repeatedly connecting and disconnecting. (freedv-backend PR #44)
+    * FreeDV Reporter: fix exiting FreeDV taking up to 60 seconds when the server stalls setting up the secure connection; give up on it after 10 seconds instead. (freedv-backend PR #63)
+    * FreeDV Reporter: fix memory leak when FreeDV exits just as FreeDV Reporter connects. (freedv-backend PR #75)
+    * Recording: fix up to the last 100 ms of audio being lost when a recording stops, and log when audio has to be dropped. (freedv-backend PR #71, #72)
+    * Fix conversion of 16-bit audio to floating point going slightly past full scale. (freedv-backend PR #60)
+    * Fix buffer overruns when one RADE receive call decodes several frames and when resampling 8 kHz audio by large ratios. (freedv-backend PR #78)
+    * Fix the audio pipeline rebuilding its resampler when the sample rate is unchanged but not when it changes, and freeing the kept resampler twice. (freedv-backend PR #78)
+    * FreeDV Reporter: refuse a secure connection if TLS 1.2 or certificate hostname checking can't be enforced. (freedv-backend PR #78)
+    * macOS/FreeDV Reporter: fix endless loop and log spam when the connection closes while sending. (freedv-backend PR #78)
+    * Fix network and threading code continuing after a failed socket or semaphore setup, Linux semaphore waits ending early on a signal, and a macOS crash on an empty background task. (freedv-backend PR #78)
+    * Windows: fix exiting FreeDV taking 20 seconds or more while logging heavily. (freedv-backend PR #78)
 2. Build system:
     * Windows versions are now built with llvm-mingw 20260908 (PR #1489)
     * Ccache support extended to third party dependencies. (PR #1498, #1505))
@@ -1036,12 +1056,17 @@ LDPC | Low Density Parity Check Codes - a family of powerful FEC codes
     * Add unit tests for non-GUI code. (PR #1542)
     * Add tests of whole transmissions through the TX/RX threads without sound cards. (PR #1567)
     * Resolve build failure on OpenSUSE Tumbleweed. (PR #1577)
+    * Fix Windows build failures with newer llvm-mingw (clang/libc++) releases. (freedv-backend PR #51)
+    * Fix Windows build with older mingw-w64 headers that don't declare SetThreadDescription. (freedv-backend PR #60)
+    * macOS: fix RNNoise build failure with newer Xcode versions. (freedv-backend PR #59)
+    * Fix build errors with GCC 16 and with OpenSSL 4 deprecations. (freedv-backend PR #41, #73)
 3. Documentation:
     * Update documentation to reflect new repo URL. (PR #1569)
 4. Other:
     * Various GUI performance improvements. (PR #1481, #1521, #1524, #1527, #1531)
     * Windows audio thread timing improvements. (PR #1488)
     * Log when the sound server puts a PulseAudio stream on another device. (PR #1591)
+    * Log from the audio threads without blocking them. (freedv-backend PR #60)
 
 ## V2.4.0 August 2026
 

src/gui/dialogs/freedv_reporter.cpp

@@ -1815,6 +1815,15 @@ void FreeDVReporterDialog::OnLeftClickTooltip(wxMouseEvent& event)
 
 void FreeDVReporterDialog::OnSetFocus(wxFocusEvent& event)
 {
+    // Only select the item under the mouse if the focus came from clicking
+    // on it, not from e.g. the window being brought to the front or a
+    // tooltip closing while the mouse happens to be over the list.
+    if (!wxGetMouseState().ButtonIsDown(wxMOUSE_BTN_ANY))
+    {
+        event.Skip();
+        return;
+    }
+
     CallAfter([this]() {
         const wxPoint pt = wxGetMousePosition();
         int mouseX = pt.x - m_listSpots->GetScreenPosition().x;

src/gui/dialogs/freedv_reporter.h

@@ -527,7 +527,9 @@ class FreeDVReporterDialog : public wxFrame
             
         };
 
-        bool isSelectionPossible_;
+        // Linux workaround to avoid inadvertent selections: set once the
+        // pointer has been over a row.
+        bool isSelectionPossible_ = false;
 
         void createColumn_(int col, bool visible);
         wxDataViewColumn* getColumnForModelColId_(unsigned int col);

src/main.cpp

@@ -4106,47 +4106,78 @@ bool MainFrame::validateSoundCardSetup(bool silent)
     }
     else
     {
-        const int MIN_SAMPLE_RATE_ANALOG = 16000;
-        const int MIN_SAMPLE_RATE_RADIO = 8000;
-        int failedSampleRate = 0;
-        int expectedSampleRate = 0;
-        int expectedSampleRate1Out = (g_nSoundCards.load(std::memory_order_acquire) == 1 ? MIN_SAMPLE_RATE_ANALOG : MIN_SAMPLE_RATE_RADIO);
-        
-        // Validate sample rates
-        if (wxGetApp().appConfiguration.audioConfiguration.soundCard1In.deviceName != "none" && wxGetApp().appConfiguration.audioConfiguration.soundCard1In.sampleRate < MIN_SAMPLE_RATE_RADIO)
+        // Below 8 kHz, FreeDV can't run at all. Mic and speaker devices
+        // below 16 kHz still work (audio is resampled), but they can't carry
+        // the upper half of RADE's 8 kHz audio bandwidth, so warn instead.
+        const int MIN_SAMPLE_RATE = 8000;
+        const int MIN_SAMPLE_RATE_ANALOG_RECOMMENDED = 16000;
+        auto& audioConfig = wxGetApp().appConfiguration.audioConfiguration;
+        bool oneSoundCard = g_nSoundCards.load(std::memory_order_acquire) == 1;
+
+        struct DeviceToCheck
         {
-            failedDeviceName = wxGetApp().appConfiguration.audioConfiguration.soundCard1In.deviceName.get();
-            failedSampleRate = wxGetApp().appConfiguration.audioConfiguration.soundCard1In.sampleRate;
-            expectedSampleRate = MIN_SAMPLE_RATE_RADIO;
-            canRun = false;
+            wxString deviceName;
+            int sampleRate;
+            bool isAnalog;
+        };
+        DeviceToCheck devicesToCheck[] = {
+            { audioConfig.soundCard1In.deviceName, audioConfig.soundCard1In.sampleRate, false },
+            { audioConfig.soundCard1Out.deviceName, audioConfig.soundCard1Out.sampleRate, oneSoundCard },
+            { audioConfig.soundCard2In.deviceName, audioConfig.soundCard2In.sampleRate, true },
+            { audioConfig.soundCard2Out.deviceName, audioConfig.soundCard2Out.sampleRate, true },
+        };
+
+        std::vector<wxString> lowRateDeviceList;
+        wxString lowRateAnalogDevices;
+        for (auto& device : devicesToCheck)
+        {
+            const wxString& deviceName = device.deviceName;
+            if (deviceName == "none")
+            {
+                continue;
+            }
+
+            if (device.sampleRate < MIN_SAMPLE_RATE)
+            {
+                if (!silent)
+                {
+                    wxMessageBox(wxString::Format(
+                        "Your %s device is set to use a sample rate of %d, which is less than the minimum of %d. Please go to Tools->Audio Config... to check your settings.",
+                        deviceName, device.sampleRate, MIN_SAMPLE_RATE), wxT("Sample Rate Too Low"), wxOK, this);
+                }
+                canRun = false;
+                break;
+            }
+            else if (device.isAnalog && device.sampleRate < MIN_SAMPLE_RATE_ANALOG_RECOMMENDED)
+            {
+                log_warn("%s is using a sample rate of %d; audio above %d Hz will be lost", (const char*)deviceName.ToUTF8(), device.sampleRate, device.sampleRate / 2);
+                wxString lowRateDeviceString = wxString::Format("%s (%d Hz)", deviceName, device.sampleRate);
+                bool found = false;
+                for (auto& str : lowRateDeviceList)
+                {
+                    if (str == lowRateDeviceString)
+                    {
+                        found = true;
+                        break;
+                    }
+                }
+                if (!found) lowRateDeviceList.push_back(lowRateDeviceString);
+            }
         }
-        else if (wxGetApp().appConfiguration.audioConfiguration.soundCard1Out.deviceName != "none" && wxGetApp().appConfiguration.audioConfiguration.soundCard1Out.sampleRate < expectedSampleRate1Out)
+
+        // Only warn once per session so that each Start doesn't bring this up again.
+        for (auto& str : lowRateDeviceList)
         {
-            failedDeviceName = wxGetApp().appConfiguration.audioConfiguration.soundCard1Out.deviceName.get();
-            failedSampleRate = wxGetApp().appConfiguration.audioConfiguration.soundCard1Out.sampleRate;
-            expectedSampleRate = expectedSampleRate1Out;
-            canRun = false;
+            lowRateAnalogDevices += wxString::Format("\n%s", str);
         }
-        else if (wxGetApp().appConfiguration.audioConfiguration.soundCard2In.deviceName != "none" && wxGetApp().appConfiguration.audioConfiguration.soundCard2In.sampleRate < MIN_SAMPLE_RATE_ANALOG)
-        {
-            failedDeviceName = wxGetApp().appConfiguration.audioConfiguration.soundCard2In.deviceName.get();
-            failedSampleRate = wxGetApp().appConfiguration.audioConfiguration.soundCard2In.sampleRate;
-            expectedSampleRate = MIN_SAMPLE_RATE_ANALOG;
-            canRun = false;
-        }
-        else if (wxGetApp().appConfiguration.audioConfiguration.soundCard2Out.deviceName != "none" && wxGetApp().appConfiguration.audioConfiguration.soundCard2Out.sampleRate < MIN_SAMPLE_RATE_ANALOG)
-        {
-            failedDeviceName = wxGetApp().appConfiguration.audioConfiguration.soundCard2Out.deviceName.get();
-            failedSampleRate = wxGetApp().appConfiguration.audioConfiguration.soundCard2Out.sampleRate;
-            expectedSampleRate = MIN_SAMPLE_RATE_ANALOG;
-            canRun = false;
-        }
-        
-        if (!canRun && !silent)
+
+        static bool lowRateWarningShown = false;
+        if (canRun && !silent && !lowRateAnalogDevices.IsEmpty() && !lowRateWarningShown)
         {
+            lowRateWarningShown = true;
             wxMessageBox(wxString::Format(
-                "Your %s device is set to use a sample rate of %d, which is less than the minimum of %d. Please go to Tools->Audio Config... to check your settings.",
-                failedDeviceName, failedSampleRate, expectedSampleRate), wxT("Sample Rate Too Low"), wxOK, this);
+                "The following microphone/speaker devices are set to a sample rate below the recommended minimum of %d Hz:\n%s\n\nFreeDV will still work, but speech will sound muffled to you or to other stations, depending on which device it is. If your device supports a sample rate higher than 16 kHz, consider enabling it.",
+                MIN_SAMPLE_RATE_ANALOG_RECOMMENDED, lowRateAnalogDevices), wxT("Low Sample Rate"), wxOK | wxICON_WARNING, this);
         }
     }
     

src/pipeline/test/RecordStepTest.cpp

@@ -147,15 +147,24 @@ bool recordsSamplesInOrder()
 {
     Recording recording;
 
-    // Pace things the way the audio device would so that this is about
-    // ordering rather than how much can be queued at once.
+    // Let each burst reach the file before queuing the next so that this is
+    // about ordering rather than how much can be queued at once (a fixed sleep
+    // isn't enough when the writer thread is starved, as on Windows runners).
     std::vector<short> expected;
     for (int block = 0; block < 100; block++)
     {
         auto samples = ramp(BLOCK_SIZE, block * BLOCK_SIZE);
         expected.insert(expected.end(), samples.begin(), samples.end());
         recording.record(samples);
-        if (block % 10 == 9) std::this_thread::sleep_for(std::chrono::milliseconds(5));
+        if (block % 10 == 9)
+        {
+            auto deadline = std::chrono::steady_clock::now() + std::chrono::seconds(5);
+            while (recording.numSamplesReported < (int)expected.size() &&
+                   std::chrono::steady_clock::now() < deadline)
+            {
+                std::this_thread::sleep_for(std::chrono::milliseconds(1));
+            }
+        }
     }
 
     return samplesAre(recording.finish(), expected);