Moved the AGC render sample queue into the audio processing module

webrtc/modules/audio_processing/gain_control_unittest.cc

 /*
  *  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.
  */
 #include <vector>
@@ skipping 12 matching lines @@
 
 void ProcessOneFrame(int sample_rate_hz,
                      AudioBuffer* render_audio_buffer,
                      AudioBuffer* capture_audio_buffer,
                      GainControlImpl* gain_controller) {
   if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
     render_audio_buffer->SplitIntoFrequencyBands();
     capture_audio_buffer->SplitIntoFrequencyBands();
   }
 
-  gain_controller->ProcessRenderAudio(render_audio_buffer);
-  gain_controller->ReadQueuedRenderData();
+  std::vector<int16_t> render_audio;
+  GainControlImpl::PackRenderAudioBuffer(render_audio_buffer, &render_audio);
+  gain_controller->ProcessRenderAudio(render_audio);
   gain_controller->AnalyzeCaptureAudio(capture_audio_buffer);
   gain_controller->ProcessCaptureAudio(capture_audio_buffer, false);
 
   if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
     capture_audio_buffer->MergeFrequencyBands();
   }
 }
 
 void SetupComponent(int sample_rate_hz,
                     GainControl::Mode mode,
@@ skipping 386 matching lines @@
      DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
 #endif
   const int kStreamAnalogLevelReference = 100;
   const float kOutputReference[] = {-0.005859f, -0.004120f, -0.002594f};
   RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100,
                       30, true, 0, 100, kStreamAnalogLevelReference,
                       kOutputReference);
 }
 
 }  // namespace webrtc