Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API

webrtc/modules/audio_processing/test/process_test.cc

 /*
  *  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 659 matching lines @@
                        sizeof(reverse_cb->channels()[i][0]));
           }
         }
 
         if (perf_testing) {
           t0 = TickTime::Now();
         }
 
         if (msg.has_data()) {
           ASSERT_EQ(apm->kNoError,
-                    apm->AnalyzeReverseStream(&far_frame));
+                    apm->ProcessReverseStream(&far_frame));
         } else {
           ASSERT_EQ(apm->kNoError,
                     apm->AnalyzeReverseStream(
                         reverse_cb->channels(),
                         far_frame.samples_per_channel_,
                         far_frame.sample_rate_hz_,
                         LayoutFromChannels(far_frame.num_channels_)));
         }
 
         if (perf_testing) {
@@ skipping 227 matching lines @@
           }
         } else {
           ASSERT_EQ(size, read_count);
         }
 
         if (perf_testing) {
           t0 = TickTime::Now();
         }
 
         ASSERT_EQ(apm->kNoError,
-                  apm->AnalyzeReverseStream(&far_frame));
+                  apm->ProcessReverseStream(&far_frame));
 
         if (perf_testing) {
           t1 = TickTime::Now();
           TickInterval tick_diff = t1 - t0;
           acc_ticks += tick_diff;
           if (tick_diff.Microseconds() > max_time_reverse_us) {
             max_time_reverse_us = tick_diff.Microseconds();
           }
           if (tick_diff.Microseconds() < min_time_reverse_us) {
             min_time_reverse_us = tick_diff.Microseconds();
@@ skipping 207 matching lines @@
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
 
   // Optional, but removes memory leak noise from Valgrind.
   google::protobuf::ShutdownProtobufLibrary();
   return 0;
 }