diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
index c2983fc625..266e649181 100644
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -63,7 +63,8 @@ bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
 }  // namespace
 
 void AecDumpBasedSimulator::PrepareProcessStreamCall(
-    const webrtc::audioproc::Stream& msg) {
+    const webrtc::audioproc::Stream& msg,
+    bool* set_stream_analog_level_called) {
   if (msg.has_input_data()) {
     // Fixed interface processing.
     // Verify interface invariance.
@@ -127,6 +128,9 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall(
   if (msg.has_level()) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->gain_control()->set_stream_analog_level(msg.level()));
+    *set_stream_analog_level_called = true;
+  } else {
+    *set_stream_analog_level_called = false;
   }
 }
 
@@ -507,8 +511,14 @@ void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) {
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::Stream& msg) {
-  PrepareProcessStreamCall(msg);
+  bool set_stream_analog_level_called = false;
+  PrepareProcessStreamCall(msg, &set_stream_analog_level_called);
   ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
+  if (set_stream_analog_level_called) {
+    // Call stream analog level to ensure that any side-effects are triggered.
+    (void)ap_->gain_control()->stream_analog_level();
+  }
+
   VerifyProcessStreamBitExactness(msg);
 }
 
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
index c3d273cf4a..7f98f43b66 100644
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -42,7 +42,8 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
   void HandleMessage(const webrtc::audioproc::Stream& msg);
   void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
   void HandleMessage(const webrtc::audioproc::Config& msg);
-  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg,
+                                bool* set_stream_analog_level_called);
   void PrepareReverseProcessStreamCall(
       const webrtc::audioproc::ReverseStream& msg);
   void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
diff --git a/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
index 673b274f72..dd680dfdfb 100644
--- a/webrtc/modules/audio_processing/test/wav_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
@@ -100,6 +100,8 @@ bool WavBasedSimulator::HandleProcessStreamCall() {
   if (samples_left_to_process) {
     PrepareProcessStreamCall();
     ProcessStream(settings_.fixed_interface);
+    // Call stream analog level to ensure that any side-effects are triggered.
+    (void)ap_->gain_control()->stream_analog_level();
     last_specified_microphone_level_ =
         ap_->gain_control()->stream_analog_level();
   }