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

webrtc/modules/audio_processing/test/audio_processing_simulator.h

 /*
  *  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.
  */
 
@@ skipping 62 matching lines @@
   rtc::Optional<int> ns_level;
   rtc::Optional<bool> use_refined_adaptive_filter;
   bool report_performance = false;
   bool report_bitexactness = false;
   bool use_verbose_logging = false;
   bool discard_all_settings_in_aecdump = true;
   rtc::Optional<std::string> aec_dump_input_filename;
   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.
 struct TickIntervalStats {
   TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
   int64_t sum;
   int64_t max;
   int64_t min;
 };
 
@@ skipping 80 matching lines @@
   TickIntervalStats proc_time_;
   std::ofstream residual_echo_likelihood_graph_writer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_