Major AEC3 render pipeline changes

webrtc/modules/audio_processing/aec3/aec_state_unittest.cc

 /*
  *  Copyright (c) 2017 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 "webrtc/modules/audio_processing/aec3/aec_state.h"
 
+// TODO(peah): Reactivate once the next CL has landed.
+#if 0
+
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 
 // Verify the general functionality of AecState
 TEST(AecState, NormalUsage) {
   ApmDataDumper data_dumper(42);
   AecState state;
   FftBuffer X_buffer(Aec3Optimization::kNone, 30, std::vector<size_t>(1, 30));
@@ skipping 244 matching lines @@
   }
 
   // Verify that the externally reported delay is properly unset when it is no
   // longer present.
   state.Update(frequency_response, rtc::Optional<size_t>(), X_buffer, E2_main,
                E2_shadow, Y2, x, EchoPathVariability(false, false), false);
   EXPECT_FALSE(state.ExternalDelay());
 }
 
 }  // namespace webrtc
+
+#endif