Added functionality for simulating custom orders of the stream API calls in audioproc_f

webrtc/modules/audio_processing/test/wav_based_simulator.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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;
   return call_chain;
 }
 
 void WavBasedSimulator::PrepareProcessStreamCall() {
   if (settings_.fixed_interface) {
     CopyToAudioFrame(*in_buf_, &fwd_frame_);
   }
@@ skipping 16 matching lines @@
     CopyToAudioFrame(*reverse_in_buf_, &rev_frame_);
   }
 }
 
 void WavBasedSimulator::Process() {
   std::unique_ptr<test::TraceToStderr> trace_to_stderr;
   if (settings_.use_verbose_logging) {
     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();
 
   bool samples_left_to_process = true;
   int call_chain_index = 0;
   int num_forward_chunks_processed = 0;
   const int kOneBykChunksPerSecond =
       1.f / AudioProcessingSimulator::kChunksPerSecond;
   while (samples_left_to_process) {
@@ skipping 81 matching lines @@
   }
 
   SetupBuffersConfigsOutputs(
       input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
       reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
       reverse_num_channels, reverse_output_num_channels);
 }
 
 }  // namespace test
 }  // namespace webrtc