Removed TimeScheduler, Process() and TimeToNextProcess() from mixer

webrtc/modules/audio_mixer/source/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_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
 #include <memory>
 
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
-#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
 struct FrameAndMuteInfo {
   FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
   AudioFrame* frame;
   bool muted;
@@ skipping 28 matching lines @@
  public:
   // AudioProcessing only accepts 10 ms frames.
   enum { kProcessPeriodicityInMs = 10 };
 
   NewAudioConferenceMixerImpl(int id);
   ~NewAudioConferenceMixerImpl();
 
   // Must be called after ctor.
   bool Init();
 
-  // Module functions
-  int64_t TimeUntilNextProcess() override;
-  void Process() override;
-
   // NewAudioConferenceMixer functions
   void Mix(AudioFrame*) override;
   int32_t SetMixabilityStatus(MixerAudioSource* participant,
                               bool mixable) override;
   bool MixabilityStatus(const MixerAudioSource& participant) const override;
   int32_t SetMinimumMixingFrequency(Frequency freq) override;
   int32_t SetAnonymousMixabilityStatus(MixerAudioSource* participant,
                                        bool mixable) override;
   bool AnonymousMixabilityStatus(
       const MixerAudioSource& participant) const override;
@@ skipping 77 matching lines @@
   // Always mixed, anonomously.
   MixerAudioSourceList _additionalParticipantList;
 
   size_t _numMixedParticipants;
   // Determines if we will use a limiter for clipping protection during
   // mixing.
   bool use_limiter_;
 
   uint32_t _timeStamp;
 
-  // Metronome class.
-  TimeScheduler _timeScheduler;
-
-  // Counter keeping track of concurrent calls to process.
-  // Note: should never be higher than 1 or lower than 0.
-  int16_t _processCalls;
-
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_