Fix Chromium clang plugin warnings

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 260 matching lines @@
 int AudioProcessingImpl::MaybeInitializeRender(
     const ProcessingConfig& processing_config) {
   return MaybeInitialize(processing_config);
 }
 
 int AudioProcessingImpl::MaybeInitializeCapture(
     const ProcessingConfig& processing_config) {
   return MaybeInitialize(processing_config);
 }
 
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+AudioProcessingImpl::ApmDebugDumpThreadState::ApmDebugDumpThreadState()
+    : event_msg(new audioproc::Event()) {}
+
+AudioProcessingImpl::ApmDebugDumpThreadState::~ApmDebugDumpThreadState() {}
+
+AudioProcessingImpl::ApmDebugDumpState::ApmDebugDumpState()
+    : debug_file(FileWrapper::Create()) {}
+
+AudioProcessingImpl::ApmDebugDumpState::~ApmDebugDumpState() {}
+
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
 // Calls InitializeLocked() if any of the audio parameters have changed from
 // their current values (needs to be called while holding the crit_render_lock).
 int AudioProcessingImpl::MaybeInitialize(
     const ProcessingConfig& processing_config) {
   // Called from both threads. Thread check is therefore not possible.
   if (processing_config == formats_.api_format) {
     return kNoError;
   }
 
   rtc::CritScope cs_capture(&crit_capture_);
@@ skipping 1226 matching lines @@
   debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
   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
 
+AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
+    bool transient_suppressor_enabled,
+    const std::vector<Point>& array_geometry,
+    SphericalPointf target_direction)
+    : aec_system_delay_jumps(-1),
+      delay_offset_ms(0),
+      was_stream_delay_set(false),
+      last_stream_delay_ms(0),
+      last_aec_system_delay_ms(0),
+      stream_delay_jumps(-1),
+      output_will_be_muted(false),
+      key_pressed(false),
+      transient_suppressor_enabled(transient_suppressor_enabled),
+      array_geometry(array_geometry),
+      target_direction(target_direction),
+      fwd_proc_format(kSampleRate16kHz),
+      split_rate(kSampleRate16kHz) {}
+
+AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
+
+AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
+
+AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
+
 }  // namespace webrtc