Removed the memory pool from the 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 95 matching lines @@
     return NULL;
   }
   return mixer;
 }
 
 NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
     : _id(id),
       _minimumMixingFreq(kLowestPossible),
       _outputFrequency(kDefaultFrequency),
       _sampleSize(0),
-      _audioFramePool(NULL),
       _participantList(),
       _additionalParticipantList(),
       _numMixedParticipants(0),
       use_limiter_(true),
       _timeStamp(0),
       mix_calls_(0) {}
 
 bool NewAudioConferenceMixerImpl::Init() {
   _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (_crit.get() == NULL)
     return false;
 
   _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
   if (_cbCrit.get() == NULL)
     return false;
 
   Config config;
   config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
   _limiter.reset(AudioProcessing::Create(config));
   if (!_limiter.get())
     return false;
 
-  MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool,
-                                           DEFAULT_AUDIO_FRAME_POOLSIZE);
-  if (_audioFramePool == NULL)
-    return false;
-
   if (SetOutputFrequency(kDefaultFrequency) == -1)
     return false;
 
   if (_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
       _limiter->kNoError)
     return false;
 
   // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
   // divide-by-2 but -7 is used instead to give a bit of headroom since the
   // AGC is not a hard limiter.
   if (_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
     return false;
 
   if (_limiter->gain_control()->set_compression_gain_db(0) !=
       _limiter->kNoError)
     return false;
 
   if (_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
     return false;
 
   if (_limiter->gain_control()->Enable(true) != _limiter->kNoError)
     return false;
 
   return true;
 }
 
-NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {
-  MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
-  assert(_audioFramePool == NULL);
-}
-
 void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
   size_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants;
   {
     CriticalSectionScoped cs(_crit.get());
     RTC_DCHECK(mix_calls_ == 0);
     mix_calls_++;
   }
 
   AudioFrameList mixList;
   AudioFrameList rampOutList;
@@ skipping 282 matching lines @@
        participant != _participantList.end(); ++participant) {
     // Stop keeping track of passive participants if there are already
     // enough participants available (they wont be mixed anyway).
     bool mustAddToPassiveList =
         (*maxAudioFrameCounter >
          (activeList.size() + passiveWasMixedList.size() +
           passiveWasNotMixedList.size()));
 
     bool wasMixed = false;
     wasMixed = (*participant)->_mixHistory->WasMixed();
-    AudioFrame* audioFrame = NULL;
-    if (_audioFramePool->PopMemory(audioFrame) == -1) {
-      WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
-                   "failed PopMemory() call");
-      assert(false);
-      return;
-    }
+    AudioFrame* audioFrame = (*participant)->GetFramePointer();
     audioFrame->sample_rate_hz_ = _outputFrequency;
 
     auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
     if (ret == MixerAudioSource::AudioFrameInfo::kError) {
-      WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
-                   "failed to GetAudioFrameWithMuted() from participant");
-      _audioFramePool->PushMemory(audioFrame);
       continue;
     }
     const bool muted = (ret == MixerAudioSource::AudioFrameInfo::kMuted);
     if (_participantList.size() != 1) {
       // TODO(wu): Issue 3390, add support for multiple participants case.
       audioFrame->ntp_time_ms_ = -1;
     }
 
     // TODO(henrike): this assert triggers in some test cases where SRTP is
     // used which prevents NetEQ from making a VAD. Temporarily disable this
@@ skipping 46 matching lines @@
           (*mixParticipantList)[audioFrame->id_] = *participant;
           assert(mixParticipantList->size() <=
                  kMaximumAmountOfMixedParticipants);
 
           if (replaceWasMixed) {
             if (!replaceFrame.muted) {
               RampOut(*replaceFrame.frame);
             }
             rampOutList->push_back(replaceFrame);
             assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
-          } else {
-            _audioFramePool->PushMemory(replaceFrame.frame);
           }
         } else {
           if (wasMixed) {
             if (!muted) {
               RampOut(*audioFrame);
             }
             rampOutList->push_back(FrameAndMuteInfo(audioFrame, muted));
             assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
-          } else {
-            _audioFramePool->PushMemory(audioFrame);
           }
         }
       } else {
         activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
         (*mixParticipantList)[audioFrame->id_] = *participant;
         assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
       }
     } else {
       if (wasMixed) {
         ParticipantFrameStruct* part_struct =
             new ParticipantFrameStruct(*participant, audioFrame, muted);
         passiveWasMixedList.push_back(part_struct);
       } else if (mustAddToPassiveList) {
         if (!muted) {
           RampIn(*audioFrame);
         }
         ParticipantFrameStruct* part_struct =
             new ParticipantFrameStruct(*participant, audioFrame, muted);
         passiveWasNotMixedList.push_back(part_struct);
-      } else {
-        _audioFramePool->PushMemory(audioFrame);
       }
     }
   }
   assert(activeList.size() <= *maxAudioFrameCounter);
   // At this point it is known which participants should be mixed. Transfer
   // this information to this functions output parameters.
   for (AudioFrameList::const_iterator iter = activeList.begin();
        iter != activeList.end(); ++iter) {
     mixList->push_back(*iter);
   }
   activeList.clear();
   // Always mix a constant number of AudioFrames. If there aren't enough
   // active participants mix passive ones. Starting with those that was mixed
   // last iteration.
   for (ParticipantFrameStructList::const_iterator iter =
            passiveWasMixedList.begin();
        iter != passiveWasMixedList.end(); ++iter) {
     if (mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
       mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame, (*iter)->muted));
       (*mixParticipantList)[(*iter)->audioFrame->id_] = (*iter)->participant;
       assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
-    } else {
-      _audioFramePool->PushMemory((*iter)->audioFrame);
     }
     delete *iter;
   }
   // And finally the ones that have not been mixed for a while.
   for (ParticipantFrameStructList::const_iterator iter =
            passiveWasNotMixedList.begin();
        iter != passiveWasNotMixedList.end(); ++iter) {
     if (mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
       mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame, (*iter)->muted));
       (*mixParticipantList)[(*iter)->audioFrame->id_] = (*iter)->participant;
       assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
-    } else {
-      _audioFramePool->PushMemory((*iter)->audioFrame);
     }
     delete *iter;
   }
   assert(*maxAudioFrameCounter + mixListStartSize >= mixList->size());
   *maxAudioFrameCounter += mixListStartSize - mixList->size();
 }
 
 void NewAudioConferenceMixerImpl::GetAdditionalAudio(
     AudioFrameList* additionalFramesList) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                "GetAdditionalAudio(additionalFramesList)");
   // The GetAudioFrameWithMuted() callback may result in the participant being
   // removed from additionalParticipantList_. If that happens it will
   // invalidate any iterators. Create a copy of the participants list such
   // that the list of participants can be traversed safely.
   MixerAudioSourceList additionalParticipantList;
   additionalParticipantList.insert(additionalParticipantList.begin(),
                                    _additionalParticipantList.begin(),
                                    _additionalParticipantList.end());
 
   for (MixerAudioSourceList::const_iterator participant =
            additionalParticipantList.begin();
        participant != additionalParticipantList.end(); ++participant) {
-    AudioFrame* audioFrame = NULL;
-    if (_audioFramePool->PopMemory(audioFrame) == -1) {
-      WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
-                   "failed PopMemory() call");
-      assert(false);
-      return;
-    }
+    AudioFrame* audioFrame = (*participant)->GetFramePointer();
     audioFrame->sample_rate_hz_ = _outputFrequency;
     auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
     if (ret == MixerAudioSource::AudioFrameInfo::kError) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
                    "failed to GetAudioFrameWithMuted() from participant");
-      _audioFramePool->PushMemory(audioFrame);
       continue;
     }
     if (audioFrame->samples_per_channel_ == 0) {
       // Empty frame. Don't use it.
-      _audioFramePool->PushMemory(audioFrame);
       continue;
     }
     additionalFramesList->push_back(FrameAndMuteInfo(
         audioFrame, ret == MixerAudioSource::AudioFrameInfo::kMuted));
   }
 }
 
 void NewAudioConferenceMixerImpl::UpdateMixedStatus(
     const std::map<int, MixerAudioSource*>& mixedParticipantsMap) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
@@ skipping 15 matching lines @@
       }
     }
     (*participant)->_mixHistory->SetIsMixed(isMixed);
   }
 }
 
 void NewAudioConferenceMixerImpl::ClearAudioFrameList(
     AudioFrameList* audioFrameList) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                "ClearAudioFrameList(audioFrameList)");
-  for (AudioFrameList::iterator iter = audioFrameList->begin();
-       iter != audioFrameList->end(); ++iter) {
-    _audioFramePool->PushMemory(iter->frame);
-  }
   audioFrameList->clear();
 }
 
 bool NewAudioConferenceMixerImpl::IsParticipantInList(
     const MixerAudioSource& participant,
     const MixerAudioSourceList& participantList) const {
   WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                "IsParticipantInList(participant,participantList)");
   for (MixerAudioSourceList::const_iterator iter = participantList.begin();
        iter != participantList.end(); ++iter) {
@@ skipping 116 matching lines @@
 
   if (error != _limiter->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "Error from AudioProcessing: %d", error);
     assert(false);
     return false;
   }
   return true;
 }
 }  // namespace webrtc