diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index cd23cd7dfc..6e9f584c75 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -540,6 +540,7 @@ if (rtc_include_tests) {
         "../../rtc_base:rtc_base_approved",
         "../../rtc_base:rtc_json",
         "../../rtc_base:task_queue_for_test",
+        "../../rtc_base/system:file_wrapper",
         "../../system_wrappers",
         "../../test:test_support",
         "aec_dump",
diff --git a/modules/audio_processing/test/wav_based_simulator.cc b/modules/audio_processing/test/wav_based_simulator.cc
index d119f4cebf..7179fc3431 100644
--- a/modules/audio_processing/test/wav_based_simulator.cc
+++ b/modules/audio_processing/test/wav_based_simulator.cc
@@ -16,6 +16,7 @@
 
 #include "modules/audio_processing/test/test_utils.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/system/file_wrapper.h"
 
 namespace webrtc {
 namespace test {
@@ -23,13 +24,14 @@ namespace test {
 std::vector<WavBasedSimulator::SimulationEventType>
 WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
   std::vector<WavBasedSimulator::SimulationEventType> call_chain;
-  FILE* stream = OpenFile(filename.c_str(), "r");
+  FileWrapper file_wrapper = FileWrapper::OpenReadOnly(filename.c_str());
 
-  RTC_CHECK(stream) << "Could not open the custom call order file, reverting "
-                       "to using the default call order";
+  RTC_CHECK(file_wrapper.is_open())
+      << "Could not open the custom call order file, reverting "
+         "to using the default call order";
 
   char c;
-  size_t num_read = fread(&c, sizeof(char), 1, stream);
+  size_t num_read = file_wrapper.Read(&c, sizeof(char));
   while (num_read > 0) {
     switch (c) {
       case 'r':
@@ -43,14 +45,12 @@ WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
       default:
         FATAL() << "Incorrect custom call order file, reverting to using the "
                    "default call order";
-        fclose(stream);
         return WavBasedSimulator::GetDefaultEventChain();
     }
 
-    num_read = fread(&c, sizeof(char), 1, stream);
+    num_read = file_wrapper.Read(&c, sizeof(char));
   }
 
-  fclose(stream);
   return call_chain;
 }
 
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 3c237df0d3..0aa1e676d1 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -391,6 +391,18 @@ class LogCall final {
   }
 };
 
+// This class is used to explicitly ignore values in the conditional
+// logging macros.  This avoids compiler warnings like "value computed
+// is not used" and "statement has no effect".
+class LogMessageVoidify {
+ public:
+  LogMessageVoidify() = default;
+  // This has to be an operator with a precedence lower than << but
+  // higher than ?:
+  template <typename... Ts>
+  void operator&(LogStreamer<Ts...>&& streamer) {}
+};
+
 }  // namespace webrtc_logging_impl
 
 // Direct use of this class is deprecated; please use the logging macros
@@ -660,9 +672,10 @@ inline const char* AdaptString(const std::string& str) {
 #define RTC_DLOG_V(sev) RTC_LOG_V(sev)
 #define RTC_DLOG_F(sev) RTC_LOG_F(sev)
 #else
-#define RTC_DLOG_EAT_STREAM_PARAMS() \
-  while (false)                      \
-  ::rtc::webrtc_logging_impl::LogStreamer<>()
+#define RTC_DLOG_EAT_STREAM_PARAMS()                \
+  while (false)                                     \
+  ::rtc::webrtc_logging_impl::LogMessageVoidify() & \
+      (::rtc::webrtc_logging_impl::LogStreamer<>())
 #define RTC_DLOG(sev) RTC_DLOG_EAT_STREAM_PARAMS()
 #define RTC_DLOG_V(sev) RTC_DLOG_EAT_STREAM_PARAMS()
 #define RTC_DLOG_F(sev) RTC_DLOG_EAT_STREAM_PARAMS()