Reland of added functionality for specifying the initial signal level to use for the gain estimation

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ skipping 95 matching lines @@
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
   virtual int InitializeLocked()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
  private:
+  // TODO(peah): These friend classes should be removed as soon as the new
+  // parameter setting scheme allows.
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
   struct ApmPublicSubmodules;
   struct ApmPrivateSubmodules;
 
   class ApmSubmoduleStates {
    public:
     ApmSubmoduleStates();
     // Updates the submodule state and returns true if it has changed.
     bool Update(bool high_pass_filter_enabled,
                 bool echo_canceller_enabled,
                 bool mobile_echo_controller_enabled,
@@ skipping 117 matching lines @@
   rtc::CriticalSection crit_debug_;
 
   // Debug dump state.
   ApmDebugDumpState debug_dump_;
 #endif
 
   // Critical sections.
   rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection crit_capture_;
 
+  // Struct containing the Config specifying the behavior of APM.
+  AudioProcessing::Config config_;
+
   // Class containing information about what submodules are active.
   ApmSubmoduleStates submodule_states_;
 
   // Structs containing the pointers to the submodules.
   std::unique_ptr<ApmPublicSubmodules> public_submodules_;
   std::unique_ptr<ApmPrivateSubmodules> private_submodules_
       GUARDED_BY(crit_capture_);
 
   // State that is written to while holding both the render and capture locks
   // but can be read without any lock being held.
@@ skipping 69 matching lines @@
     ApmRenderState();
     ~ApmRenderState();
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_