Lock scheme #4: Introduced the render sample queue for the aec and aecm

webrtc/modules/audio_processing/echo_cancellation_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_ECHO_CANCELLATION_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
 class AudioBuffer;
 class CriticalSectionWrapper;
 
 class EchoCancellationImpl : public EchoCancellation,
                              public ProcessingComponent {
@@ skipping 11 matching lines @@
   SuppressionLevel suppression_level() const override;
   bool is_drift_compensation_enabled() const override;
 
   // ProcessingComponent implementation.
   int Initialize() override;
   void SetExtraOptions(const Config& config) override;
 
   bool is_delay_agnostic_enabled() const;
   bool is_extended_filter_enabled() const;
 
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
+
  private:
+  static const size_t kMaxNumChannelsPerFrameToBuffer = 2;
+  static const size_t kMinNumSamplesPerFrameToBuffer = 80;
+  static const size_t kMaxNumSamplesPerFrameToBuffer = 160;
+  // TODO(peah): Decrease this once we properly handle hugely unbalanced
+  // reverse and forward call numbers.
+  static const size_t kMaxNumFramesToBuffer = 100;
+
   // EchoCancellation implementation.
   int Enable(bool enable) override;
   int enable_drift_compensation(bool enable) override;
   void set_stream_drift_samples(int drift) override;
   int set_suppression_level(SuppressionLevel level) override;
   int enable_metrics(bool enable) override;
   bool are_metrics_enabled() const override;
   bool stream_has_echo() const override;
   int GetMetrics(Metrics* metrics) override;
   int enable_delay_logging(bool enable) override;
   bool is_delay_logging_enabled() const override;
   int GetDelayMetrics(int* median, int* std) override;
   int GetDelayMetrics(int* median,
                       int* std,
                       float* fraction_poor_delays) override;
   struct AecCore* aec_core() const override;
 
   // ProcessingComponent implementation.
   void* CreateHandle() const override;
   int InitializeHandle(void* handle) const override;
   int ConfigureHandle(void* handle) const override;
   void DestroyHandle(void* handle) const override;
   int num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  // SwapQueue element verifier
+  static bool RenderQueueItemVerifier(const std::vector<float>& v);
+
   const AudioProcessing* apm_;
   CriticalSectionWrapper* crit_;
   bool drift_compensation_enabled_;
   bool metrics_enabled_;
   SuppressionLevel suppression_level_;
   int stream_drift_samples_;
   bool was_stream_drift_set_;
   bool stream_has_echo_;
   bool delay_logging_enabled_;
   bool extended_filter_enabled_;
   bool delay_agnostic_enabled_;
+
+  std::vector<float> render_queue_buffer_;
+  std::vector<float> capture_queue_buffer_;
+  rtc::scoped_ptr<SwapQueue<std::vector<float>, &RenderQueueItemVerifier>>
+      render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_