Use ProcessReverseStream in VoiceEngines OutputMixer

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 424 matching lines @@
     capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
     if (config.Get<Beamforming>().array_geometry.size() > 1) {
       capture_.array_geometry = config.Get<Beamforming>().array_geometry;
     }
     capture_.target_direction = config.Get<Beamforming>().target_direction;
     InitializeBeamformer();
   }
 #endif  // WEBRTC_ANDROID_PLATFORM_BUILD
 }
 
-int AudioProcessingImpl::input_sample_rate_hz() const {
-  // Accessed from outside APM, hence a lock is needed.
-  rtc::CritScope cs(&crit_capture_);
-  return formats_.api_format.input_stream().sample_rate_hz();
-}
-
 int AudioProcessingImpl::proc_sample_rate_hz() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.split_rate;
 }
 
@@ skipping 372 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");
+  rtc::CritScope cs(&crit_render_);
   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_);
@@ skipping 587 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