Added a level indicator to new mixer.

webrtc/modules/audio_mixer/new_audio_conference_mixer_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_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/voice_engine/level_indicator.h"
 
 namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
 struct FrameAndMuteInfo {
   FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
   AudioFrame* frame;
   bool muted;
 };
@@ skipping 85 matching lines @@
                              bool use_limiter);
 
   // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
   // record will be kept of this mix (e.g. the corresponding MixerAudioSources
   // will not be marked as IsMixed()
   int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
                                  const AudioFrameList& audioFrameList) const;
 
   bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
+  // Output level functions for VoEVolumeControl.
+  int GetOutputAudioLevel() override;
+
+  int GetOutputAudioLevelFullRange() override;
+
   std::unique_ptr<CriticalSectionWrapper> crit_;
   std::unique_ptr<CriticalSectionWrapper> cb_crit_;
 
   int32_t id_;
 
   // The current sample frequency and sample size when mixing.
   Frequency output_frequency_;
   size_t sample_size_;
 
   // List of all audio sources. Note all lists are disjunct
   MixerAudioSourceList audio_source_list_;  // May be mixed.
 
   // Always mixed, anonomously.
   MixerAudioSourceList additional_audio_source_list_;
 
   size_t num_mixed_audio_sources_;
   // Determines if we will use a limiter for clipping protection during
   // mixing.
   bool use_limiter_;
 
   uint32_t time_stamp_;
 
   // Ensures that Mix is called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> limiter_;
+
+  // Measures audio level for the combined signal.
+  voe::AudioLevel audio_level_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_