Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API

webrtc/modules/audio_processing/audio_processing_impl.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 816 matching lines @@
   }
 #endif
 
   render_.render_audio->CopyFrom(src,
                                  formats_.api_format.reverse_input_stream());
   return ProcessReverseStreamLocked();
 }
 
 int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
   TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
-  RETURN_ON_ERR(AnalyzeReverseStream(frame));
-  rtc::CritScope cs(&crit_render_);
-  if (is_rev_processed()) {
-    render_.render_audio->InterleaveTo(frame, true);
-  }
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
-  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_AudioFrame");
   rtc::CritScope cs(&crit_render_);
   if (frame == nullptr) {
     return kNullPointerError;
   }
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
       frame->sample_rate_hz_ != kSampleRate32kHz &&
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
@@ skipping 26 matching lines @@
         debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_data(frame->data_, data_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &debug_dump_.num_bytes_left_for_log_,
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
   render_.render_audio->DeinterleaveFrom(frame);
-  return ProcessReverseStreamLocked();
+  RETURN_ON_ERR(ProcessReverseStreamLocked());
+  if (is_rev_processed()) {
+    render_.render_audio->InterleaveTo(frame, true);
+  }
+  return kNoError;
 }
 
 int AudioProcessingImpl::ProcessReverseStreamLocked() {
   AudioBuffer* ra = render_.render_audio.get();  // For brevity.
   if (rev_analysis_needed()) {
     ra->SplitIntoFrequencyBands();
   }
 
   if (constants_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
@@ skipping 531 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc