diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
index 2564dc4bf2..06735c47b0 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -81,6 +81,7 @@ struct SimulationSettings {
   rtc::Optional<std::string> aec_dump_output_filename;
   bool fixed_interface = false;
   bool store_intermediate_output = false;
+  rtc::Optional<std::string> custom_call_order_filename;
 };
 
 // Holds a few statistics about a series of TickIntervals.
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index edc775986f..c4e7c3dced 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -166,6 +166,7 @@ DEFINE_bool(discard_settings_in_aecdump,
 DEFINE_bool(store_intermediate_output,
             false,
             "Creates new output files after each init");
+DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
 
 void SetSettingIfSpecified(const std::string value,
                            rtc::Optional<std::string>* parameter) {
@@ -257,6 +258,8 @@ SimulationSettings CreateSettings() {
   SetSettingIfSpecified(FLAGS_stream_delay, &settings.stream_delay);
   SetSettingIfSpecified(FLAGS_stream_drift_samples,
                         &settings.stream_drift_samples);
+  SetSettingIfSpecified(FLAGS_custom_call_order_file,
+                        &settings.custom_call_order_filename);
   settings.report_performance = FLAGS_performance_report;
   settings.use_verbose_logging = FLAGS_verbose;
   settings.report_bitexactness = FLAGS_bitexactness_report;
@@ -368,6 +371,11 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
       "Error: --bitexactness_report can only be used when operating on an "
       "aecdump\n");
 
+  ReportConditionalErrorAndExit(
+      settings.custom_call_order_filename && settings.aec_dump_input_filename,
+      "Error: --custom_call_order_file cannot be used when operating on an "
+      "aecdump\n");
+
   auto valid_wav_name = [](const std::string& wav_file_name) {
     if (wav_file_name.size() < 5) {
       return false;
diff --git a/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
index dd680dfdfb..6cf0b744e6 100644
--- a/webrtc/modules/audio_processing/test/wav_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
@@ -10,19 +10,57 @@
 
 #include "webrtc/modules/audio_processing/test/wav_based_simulator.h"
 
+#include <stdio.h>
+#include <iostream>
+
 #include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
 namespace webrtc {
 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");
+
+  RTC_CHECK(stream) << "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);
+  while (num_read > 0) {
+    switch (c) {
+      case 'r':
+        call_chain.push_back(SimulationEventType::kProcessReverseStream);
+        break;
+      case 'c':
+        call_chain.push_back(SimulationEventType::kProcessStream);
+        break;
+      case '\n':
+        break;
+      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);
+  }
+
+  fclose(stream);
+  return call_chain;
+}
+
 WavBasedSimulator::WavBasedSimulator(const SimulationSettings& settings)
       : AudioProcessingSimulator(settings) {}
 
 WavBasedSimulator::~WavBasedSimulator() = default;
 
 std::vector<WavBasedSimulator::SimulationEventType>
-WavBasedSimulator::GetDefaultEventChain() const {
+WavBasedSimulator::GetDefaultEventChain() {
   std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);
   call_chain[0] = SimulationEventType::kProcessStream;
   call_chain[1] = SimulationEventType::kProcessReverseStream;
@@ -59,7 +97,12 @@ void WavBasedSimulator::Process() {
     trace_to_stderr.reset(new test::TraceToStderr(true));
   }
 
-  call_chain_ = GetDefaultEventChain();
+  if (settings_.custom_call_order_filename) {
+    call_chain_ = WavBasedSimulator::GetCustomEventChain(
+        *settings_.custom_call_order_filename);
+  } else {
+    call_chain_ = WavBasedSimulator::GetDefaultEventChain();
+  }
   CreateAudioProcessor();
 
   Initialize();
diff --git a/webrtc/modules/audio_processing/test/wav_based_simulator.h b/webrtc/modules/audio_processing/test/wav_based_simulator.h
index 5bf93d16d1..c1fcf6163b 100644
--- a/webrtc/modules/audio_processing/test/wav_based_simulator.h
+++ b/webrtc/modules/audio_processing/test/wav_based_simulator.h
@@ -40,7 +40,9 @@ class WavBasedSimulator final : public AudioProcessingSimulator {
   bool HandleProcessReverseStreamCall();
   void PrepareProcessStreamCall();
   void PrepareReverseProcessStreamCall();
-  std::vector<SimulationEventType> GetDefaultEventChain() const;
+  static std::vector<SimulationEventType> GetDefaultEventChain();
+  static std::vector<SimulationEventType> GetCustomEventChain(
+      const std::string& filename);
 
   std::vector<SimulationEventType> call_chain_;
   int last_specified_microphone_level_ = 100;