Moved the AGC render sample queue into the audio processing module

webrtc/modules/audio_processing/gain_control_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.
  */
 
@@ skipping 13 matching lines @@
 namespace webrtc {
 
 class AudioBuffer;
 
 class GainControlImpl : public GainControl {
  public:
   GainControlImpl(rtc::CriticalSection* crit_render,
                   rtc::CriticalSection* crit_capture);
   ~GainControlImpl() override;
 
-  int ProcessRenderAudio(AudioBuffer* audio);
+  void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
   int AnalyzeCaptureAudio(AudioBuffer* audio);
   int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
 
   void Initialize(size_t num_proc_channels, int sample_rate_hz);
 
+  static void PackRenderAudioBuffer(AudioBuffer* audio,
+                                    std::vector<int16_t>* packed_buffer);
+
   // GainControl implementation.
   bool is_enabled() const override;
   int stream_analog_level() override;
   bool is_limiter_enabled() const override;
   Mode mode() const override;
 
-  // Reads render side data that has been queued on the render call.
-  void ReadQueuedRenderData();
-
   int compression_gain_db() const override;
 
  private:
   class GainController;
 
   // GainControl implementation.
   int Enable(bool enable) override;
   int set_stream_analog_level(int level) override;
   int set_mode(Mode mode) override;
   int set_target_level_dbfs(int level) override;
   int target_level_dbfs() const override;
   int set_compression_gain_db(int gain) override;
   int enable_limiter(bool enable) override;
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
   int analog_level_maximum() const override;
   bool stream_is_saturated() const override;
 
-  void AllocateRenderQueue();
   int Configure();
 
   rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection* const crit_capture_;
 
   bool enabled_ = false;
 
   Mode mode_ GUARDED_BY(crit_capture_);
   int minimum_capture_level_ GUARDED_BY(crit_capture_);
   int maximum_capture_level_ GUARDED_BY(crit_capture_);
   bool limiter_enabled_ GUARDED_BY(crit_capture_);
   int target_level_dbfs_ GUARDED_BY(crit_capture_);
   int compression_gain_db_ GUARDED_BY(crit_capture_);
   int analog_capture_level_ GUARDED_BY(crit_capture_);
   bool was_analog_level_set_ GUARDED_BY(crit_capture_);
   bool stream_is_saturated_ GUARDED_BY(crit_capture_);
 
-  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
-      GUARDED_BY(crit_capture_);
-  std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
-  std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
-
-  // Lock protection not needed.
-  std::unique_ptr<
-      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
-      render_signal_queue_;
-
   std::vector<std::unique_ptr<GainController>> gain_controllers_;
 
   rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
   rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_