Add a new UMA metric in APM to track incoming capture-side audio level

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/function_view.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #endif
@@ skipping 361 matching lines @@
   size_t agc_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
   std::vector<int16_t> agc_render_queue_buffer_ GUARDED_BY(crit_render_);
   std::vector<int16_t> agc_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
   size_t red_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
   std::vector<float> red_render_queue_buffer_ GUARDED_BY(crit_render_);
   std::vector<float> red_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
+  RMSLevel rms_ GUARDED_BY(crit_capture_);
+  int rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
+
   // Lock protection not needed.
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       aec_render_signal_queue_;
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       aecm_render_signal_queue_;
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       agc_render_signal_queue_;
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       red_render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_