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

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_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 101 matching lines @@
 NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
     : _id(id),
       _minimumMixingFreq(kLowestPossible),
       _outputFrequency(kDefaultFrequency),
       _sampleSize(0),
       _audioFramePool(NULL),
       audio_source_list_(),
       additional_audio_source_list_(),
       num_mixed_audio_sources_(0),
       use_limiter_(true),
-      _timeStamp(0),
-      _timeScheduler(kProcessPeriodicityInMs),
-      _processCalls(0) {}
+      _timeStamp(0) {
+  thread_checker_.DetachFromThread();
+      }
 
 bool NewAudioConferenceMixerImpl::Init() {
   _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (_crit.get() == NULL)
     return false;
 
   _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (_cbCrit.get() == NULL)
     return false;
 
@@ skipping 32 matching lines @@
     return false;
 
   return true;
 }
 
 NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {
   MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
   RTC_DCHECK_EQ(_audioFramePool, static_cast<MemoryPool<AudioFrame>*>(nullptr));
 }
 
-// Process should be called every kProcessPeriodicityInMs ms
-int64_t NewAudioConferenceMixerImpl::TimeUntilNextProcess() {
-  int64_t timeUntilNextProcess = 0;
-  CriticalSectionScoped cs(_crit.get());
-  if (_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
-                 "failed in TimeToNextUpdate() call");
-    // Sanity check
-    RTC_NOTREACHED();
-    return -1;
-  }
-  return timeUntilNextProcess;
-}
-
-void NewAudioConferenceMixerImpl::Process() {
-  // TODO(aleloi) Remove this method.
-  RTC_NOTREACHED();
-}
-
 void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
   size_t remainingAudioSourcesAllowedToMix = kMaximumAmountOfMixedAudioSources;
-  {
-    CriticalSectionScoped cs(_crit.get());
-    RTC_DCHECK_EQ(_processCalls, 0);
-    _processCalls++;
-
-    // Let the scheduler know that we are running one iteration.
-    _timeScheduler.UpdateScheduler();
-  }
-
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   AudioFrameList mixList;
   AudioFrameList rampOutList;
   AudioFrameList additionalFramesList;
   std::map<int, MixerAudioSource*> mixedAudioSourcesMap;
   {
     CriticalSectionScoped cs(_cbCrit.get());
 
     int32_t lowFreq = GetLowestMixingFrequency();
     // SILK can run in 12 kHz and 24 kHz. These frequencies are not
     // supported so use the closest higher frequency to not lose any
     // information.
     // TODO(aleloi): this is probably more appropriate to do in
     //                GetLowestMixingFrequency().
     if (lowFreq == 12000) {
       lowFreq = 16000;
     } else if (lowFreq == 24000) {
       lowFreq = 32000;
     }
     if (lowFreq <= 0) {
-      CriticalSectionScoped cs(_crit.get());
-      _processCalls--;
       return;
     } else {
       switch (lowFreq) {
         case 8000:
           if (OutputFrequency() != kNbInHz) {
             SetOutputFrequency(kNbInHz);
           }
           break;
         case 16000:
           if (OutputFrequency() != kWbInHz) {
             SetOutputFrequency(kWbInHz);
           }
           break;
         case 32000:
           if (OutputFrequency() != kSwbInHz) {
             SetOutputFrequency(kSwbInHz);
           }
           break;
         case 48000:
           if (OutputFrequency() != kFbInHz) {
             SetOutputFrequency(kFbInHz);
           }
           break;
         default:
           RTC_NOTREACHED();
-
-          CriticalSectionScoped cs(_crit.get());
-          _processCalls--;
           return;
       }
     }
 
     UpdateToMix(&mixList, &rampOutList, &mixedAudioSourcesMap,
                 &remainingAudioSourcesAllowedToMix);
 
     GetAdditionalAudio(&additionalFramesList);
     UpdateMixedStatus(mixedAudioSourcesMap);
   }
@@ skipping 30 matching lines @@
       audio_frame_for_mixing->Mute();
     } else {
       // Only call the limiter if we have something to mix.
       LimitMixedAudio(audio_frame_for_mixing);
     }
   }
 
   ClearAudioFrameList(&mixList);
   ClearAudioFrameList(&rampOutList);
   ClearAudioFrameList(&additionalFramesList);
-  {
-    CriticalSectionScoped cs(_crit.get());
-    _processCalls--;
-  }
   return;
 }
 
 int32_t NewAudioConferenceMixerImpl::SetOutputFrequency(
     const Frequency& frequency) {
   CriticalSectionScoped cs(_crit.get());
 
   _outputFrequency = frequency;
   _sampleSize =
       static_cast<size_t>((_outputFrequency * kProcessPeriodicityInMs) / 1000);
@@ skipping 549 matching lines @@
 
   if (error != _limiter->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "Error from AudioProcessing: %d", error);
     RTC_NOTREACHED();
     return false;
   }
   return true;
 }
 }  // namespace webrtc