Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 60 matching lines @@
       int input_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       int output_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       float* const* dest));
   WEBRTC_STUB(ProcessStream,
               (const float* const* src,
                const webrtc::StreamConfig& input_config,
                const webrtc::StreamConfig& output_config,
                float* const* dest));
-  WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));
   WEBRTC_STUB(AnalyzeReverseStream, (
       const float* const* data,
       size_t samples_per_channel,
       int sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout layout));
   WEBRTC_STUB(ProcessReverseStream,
               (const float* const* src,
                const webrtc::StreamConfig& reverse_input_config,
                const webrtc::StreamConfig& reverse_output_config,
@@ skipping 682 matching lines @@
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_