Moved the AEC render sample queue into the audio processing module

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/swap_queue.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/modules/audio_processing/render_queue_item_verifier.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
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #endif
@@ skipping 191 matching lines @@
   void InitializeTransient()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeBeamformer()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeIntelligibility()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   int InitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeLevelController() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
+  void EmptyQueuedRenderAudio();
+  void AllocateRenderQueue()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void QueueRenderAudio(const AudioBuffer* audio)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
   // Capture-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   int ProcessCaptureStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   // Render-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   // TODO(ekm): Remove once all clients updated to new interface.
   int AnalyzeReverseStreamLocked(const float* const* src,
                                  const StreamConfig& input_config,
@@ skipping 109 matching lines @@
     bool intelligibility_enabled;
     bool level_controller_enabled = false;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     ApmRenderState();
     ~ApmRenderState();
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_) = 0;
+  std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
+      render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_