Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API

webrtc/modules/audio_processing/audio_processing_impl_unittest.cc

 /*
  *  Copyright (c) 2014 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 30 matching lines @@
   EXPECT_CALL(mock, InitializeLocked()).Times(1);
   mock.Initialize();
 
   AudioFrame frame;
   // Call with the default parameters; there should be no init.
   frame.num_channels_ = 1;
   SetFrameSampleRate(&frame, 16000);
   EXPECT_CALL(mock, InitializeLocked())
       .Times(0);
   EXPECT_NOERR(mock.ProcessStream(&frame));
-  EXPECT_NOERR(mock.AnalyzeReverseStream(&frame));
+  EXPECT_NOERR(mock.ProcessReverseStream(&frame));
 
   // New sample rate. (Only impacts ProcessStream).
   SetFrameSampleRate(&frame, 32000);
   EXPECT_CALL(mock, InitializeLocked())
       .Times(1);
   EXPECT_NOERR(mock.ProcessStream(&frame));
 
   // New number of channels.
   frame.num_channels_ = 2;
   EXPECT_CALL(mock, InitializeLocked())
       .Times(2);
   EXPECT_NOERR(mock.ProcessStream(&frame));
   // ProcessStream sets num_channels_ == num_output_channels.
   frame.num_channels_ = 2;
-  EXPECT_NOERR(mock.AnalyzeReverseStream(&frame));
+  EXPECT_NOERR(mock.ProcessReverseStream(&frame));
 
-  // A new sample rate passed to AnalyzeReverseStream should cause an init.
+  // A new sample rate passed to ProcessReverseStream should cause an init.
   SetFrameSampleRate(&frame, 16000);
   EXPECT_CALL(mock, InitializeLocked()).Times(1);
-  EXPECT_NOERR(mock.AnalyzeReverseStream(&frame));
+  EXPECT_NOERR(mock.ProcessReverseStream(&frame));
 }
 
 }  // namespace webrtc