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 97 matching lines @@
 NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
     : _id(id),
       _minimumMixingFreq(kLowestPossible),
       _outputFrequency(kDefaultFrequency),
       _sampleSize(0),
       _audioFramePool(NULL),
       _participantList(),
       _additionalParticipantList(),
       _numMixedParticipants(0),
       use_limiter_(true),
-      _timeStamp(0),
-      _timeScheduler(kProcessPeriodicityInMs),
-      _processCalls(0) {}
+      _timeStamp(0) {}
 
 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);
   assert(_audioFramePool == NULL);
 }
 
-// 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
-    assert(false);
-    return -1;
-  }
-  return timeUntilNextProcess;
-}
-
-void NewAudioConferenceMixerImpl::Process() {
-  RTC_NOTREACHED();
-}
-
 void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
   size_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants;
-  {
-    CriticalSectionScoped cs(_crit.get());
-    assert(_processCalls == 0);

[ivoc, 2016/07/04 13:11:52]

It seems like this code checks that this function is not called from multiple
threads, do you think it's useful to keep a check for that in the new code?

[aleloi, 2016/07/06 10:20:32]

Wow! This was important and I missed it. I thought for some reason that the mix
call counter was used to let the time scheduler know how many more calls should
be done.

I added it back in next the next patch sets.

[ivoc, 2016/07/06 14:43:26]

Although I agree that it's good to check for this, I'm not sure that the old
code is the best way to do this. In much of the rest of WebRTC code a
ThreadChecker is used for it, usually with a call to CalledOnValidThread()
inside a DCHECK (so it's only actually checked in debug builds). I don't really
see any reason not to use the same pattern here.

[aleloi, 2016/07/06 15:17:19]

Good idea! Done.

-    _processCalls++;
-
-    // Let the scheduler know that we are running one iteration.
-    _timeScheduler.UpdateScheduler();
-  }
 
   AudioFrameList mixList;
   AudioFrameList rampOutList;
   AudioFrameList additionalFramesList;
   std::map<int, MixerAudioSource*> mixedParticipantsMap;
   {
     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(henrike): 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:
           assert(false);
 
           CriticalSectionScoped cs(_crit.get());
-          _processCalls--;
           return;
       }
     }
 
     UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap,
                 &remainingParticipantsAllowedToMix);
 
     GetAdditionalAudio(&additionalFramesList);
     UpdateMixedStatus(mixedParticipantsMap);
   }
@@ skipping 31 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 535 matching lines @@
 
   if (error != _limiter->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "Error from AudioProcessing: %d", error);
     assert(false);
     return false;
   }
   return true;
 }
 }  // namespace webrtc