Preserve level controller output when no other effects are active

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 222 matching lines @@
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive()
     const {
   return low_cut_filter_enabled_ || echo_canceller_enabled_ ||
          mobile_echo_controller_enabled_ || noise_suppressor_enabled_ ||
          beamformer_enabled_ || adaptive_gain_controller_enabled_ ||
          echo_canceller3_enabled_;
 }
 
+bool AudioProcessingImpl::ApmSubmoduleStates::CaptureFullBandProcessingActive()
+    const {
+  return level_controller_enabled_;
+}
+
 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()
     const {
   return RenderMultiBandProcessingActive() || echo_canceller_enabled_ ||
          mobile_echo_controller_enabled_ || adaptive_gain_controller_enabled_ ||
          echo_canceller3_enabled_;
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandProcessingActive()
     const {
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
@@ skipping 873 matching lines @@
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_input_data(frame->data_, data_size);
   }
 #endif
 
   capture_.capture_audio->DeinterleaveFrom(frame);
   RETURN_ON_ERR(ProcessCaptureStreamLocked());
   capture_.capture_audio->InterleaveTo(
-      frame, submodule_states_.CaptureMultiBandProcessingActive());
+      frame, submodule_states_.CaptureMultiBandProcessingActive() ||
+                 submodule_states_.CaptureFullBandProcessingActive());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t data_size =
         sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
     msg->set_output_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_.capture));
@@ skipping 917 matching lines @@
       previous_agc_level(0),
       echo_path_gain_change(false) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc