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

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc

Index: webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
diff --git a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
index e71f9f0c3582e072445130201bbd5d2e41dfd93e..c08d7c532ced4554c38e3f93142e539173da0395 100644
--- a/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc
@@ -115,9 +115,7 @@ NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
       _additionalParticipantList(),
       _numMixedParticipants(0),
       use_limiter_(true),
-      _timeStamp(0),
-      _timeScheduler(kProcessPeriodicityInMs),
-      _processCalls(0) {}
+      _timeStamp(0) {}
 
 bool NewAudioConferenceMixerImpl::Init() {
   _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
@@ -170,34 +168,8 @@ NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {
   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;
@@ -219,7 +191,6 @@ void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
     }
     if (lowFreq <= 0) {
       CriticalSectionScoped cs(_crit.get());
-      _processCalls--;
       return;
     } else {
       switch (lowFreq) {
@@ -247,7 +218,6 @@ void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
           assert(false);
 
           CriticalSectionScoped cs(_crit.get());
-          _processCalls--;
           return;
       }
     }
@@ -299,10 +269,6 @@ void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
   ClearAudioFrameList(&mixList);
   ClearAudioFrameList(&rampOutList);
   ClearAudioFrameList(&additionalFramesList);
-  {
-    CriticalSectionScoped cs(_crit.get());
-    _processCalls--;
-  }
   return;
 }