Removed callback between old AudioConferenceMixer and OutputMixer. The audio frame with mixed audio…

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.cc

 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2016 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.
  */
 
 #include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
 
@@ skipping 19 matching lines @@
 
 typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList;
 
 // Mix |frame| into |mixed_frame|, with saturation protection and upmixing.
 // These effects are applied to |frame| itself prior to mixing. Assumes that
 // |mixed_frame| always has at least as many channels as |frame|. Supports
 // stereo at most.
 //
 // TODO(andrew): consider not modifying |frame| here.
 void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
-  assert(mixed_frame->num_channels_ >= frame->num_channels_);
+  RTC_DCHECK_GE(mixed_frame->num_channels_, frame->num_channels_);
   if (use_limiter) {
     // Divide by two to avoid saturation in the mixing.
     // This is only meaningful if the limiter will be used.
     *frame >>= 1;
   }
   if (mixed_frame->num_channels_ > frame->num_channels_) {
     // We only support mono-to-stereo.
-    assert(mixed_frame->num_channels_ == 2 && frame->num_channels_ == 1);
+    RTC_DCHECK_EQ(mixed_frame->num_channels_, static_cast<size_t>(2));
+    RTC_DCHECK_EQ(frame->num_channels_, static_cast<size_t>(1));
     AudioFrameOperations::MonoToStereo(frame);
   }
 
   *mixed_frame += *frame;
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
 size_t MaxNumChannels(const AudioFrameList* list) {
   size_t max_num_channels = 1;
   for (AudioFrameList::const_iterator iter = list->begin(); iter != list->end();
@@ skipping 43 matching lines @@
   if (!mixer->Init()) {
     delete mixer;
     return NULL;
   }
   return mixer;
 }
 
 NewAudioConferenceMixerImpl::NewAudioConferenceMixerImpl(int id)
     : _id(id),
       _minimumMixingFreq(kLowestPossible),
-      _mixReceiver(NULL),
       _outputFrequency(kDefaultFrequency),
       _sampleSize(0),
       _audioFramePool(NULL),
       _participantList(),
       _additionalParticipantList(),
       _numMixedParticipants(0),
       use_limiter_(true),
       _timeStamp(0),
       _timeScheduler(kProcessPeriodicityInMs),
       _processCalls(0) {}
@@ skipping 39 matching lines @@
     return false;
 
   if (_limiter->gain_control()->Enable(true) != _limiter->kNoError)
     return false;
 
   return true;
 }
 
 NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {
   MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
-  assert(_audioFramePool == NULL);
+  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
-    assert(false);
+    RTC_DCHECK(false);
     return -1;
   }
   return timeUntilNextProcess;
 }
 
 void NewAudioConferenceMixerImpl::Process() {
+  // TODO(aleloi) Remove this method.
+  RTC_NOTREACHED();
+}
+
+void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
   size_t remainingParticipantsAllowedToMix = kMaximumAmountOfMixedParticipants;
   {
     CriticalSectionScoped cs(_crit.get());
-    assert(_processCalls == 0);
+    RTC_DCHECK_EQ(_processCalls, 0);
     _processCalls++;
 
     // Let the scheduler know that we are running one iteration.
     _timeScheduler.UpdateScheduler();
   }
 
   AudioFrameList mixList;
   AudioFrameList rampOutList;
   AudioFrameList additionalFramesList;
   std::map<int, MixerAudioSource*> mixedParticipantsMap;
@@ skipping 31 matching lines @@
           if (OutputFrequency() != kSwbInHz) {
             SetOutputFrequency(kSwbInHz);
           }
           break;
         case 48000:
           if (OutputFrequency() != kFbInHz) {
             SetOutputFrequency(kFbInHz);
           }
           break;
         default:
-          assert(false);
+          RTC_DCHECK(false);

[tommi, 2016/07/08 12:24:17]

nit: RTC_NOTREACHED()

[aleloi, 2016/07/08 12:57:40]

Done.

 
           CriticalSectionScoped cs(_crit.get());
           _processCalls--;
           return;
       }
     }
 
     UpdateToMix(&mixList, &rampOutList, &mixedParticipantsMap,
                 &remainingParticipantsAllowedToMix);
 
     GetAdditionalAudio(&additionalFramesList);
     UpdateMixedStatus(mixedParticipantsMap);
   }
 
-  // Get an AudioFrame for mixing from the memory pool.
-  AudioFrame* mixedAudio = NULL;
-  if (_audioFramePool->PopMemory(mixedAudio) == -1) {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
-                 "failed PopMemory() call");
-    assert(false);
-    return;
-  }
+  // TODO(henrike): it might be better to decide the number of channels

[tommi, 2016/07/08 12:24:17]

henrike doesn't work on webrtc anymore. Can you assign this to someone else or
remove it if it's no longer relevant?

[aleloi, 2016/07/08 12:57:40]

Done.

+  //                with an API instead of dynamically.
+
+  // Find the max channels over all mixing lists.
+  const size_t num_mixed_channels = std::max(
+      MaxNumChannels(&mixList), std::max(MaxNumChannels(&additionalFramesList),
+                                         MaxNumChannels(&rampOutList)));
+
+  audio_frame_for_mixing->UpdateFrame(
+      -1, _timeStamp, NULL, 0, _outputFrequency, AudioFrame::kNormalSpeech,
+      AudioFrame::kVadPassive, num_mixed_channels);
+
+  _timeStamp += static_cast<uint32_t>(_sampleSize);
+
+  use_limiter_ = _numMixedParticipants > 1 &&
+                 _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
+
+  // We only use the limiter if it supports the output sample rate and
+  // we're actually mixing multiple streams.
+  MixFromList(audio_frame_for_mixing, mixList, _id, use_limiter_);
 
   {
     CriticalSectionScoped cs(_crit.get());
+    MixAnonomouslyFromList(audio_frame_for_mixing, additionalFramesList);
+    MixAnonomouslyFromList(audio_frame_for_mixing, rampOutList);
 
-    // TODO(henrike): it might be better to decide the number of channels
-    //                with an API instead of dynamically.
-
-    // Find the max channels over all mixing lists.
-    const size_t num_mixed_channels =
-        std::max(MaxNumChannels(&mixList),
-                 std::max(MaxNumChannels(&additionalFramesList),
-                          MaxNumChannels(&rampOutList)));
-
-    mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency,
-                            AudioFrame::kNormalSpeech, AudioFrame::kVadPassive,
-                            num_mixed_channels);
-
-    _timeStamp += static_cast<uint32_t>(_sampleSize);
-
-    // We only use the limiter if it supports the output sample rate and
-    // we're actually mixing multiple streams.
-    use_limiter_ = _numMixedParticipants > 1 &&
-                   _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
-
-    MixFromList(mixedAudio, mixList);
-    MixAnonomouslyFromList(mixedAudio, additionalFramesList);
-    MixAnonomouslyFromList(mixedAudio, rampOutList);
-
-    if (mixedAudio->samples_per_channel_ == 0) {
+    if (audio_frame_for_mixing->samples_per_channel_ == 0) {
       // Nothing was mixed, set the audio samples to silence.
-      mixedAudio->samples_per_channel_ = _sampleSize;
-      mixedAudio->Mute();
+      audio_frame_for_mixing->samples_per_channel_ = _sampleSize;
+      audio_frame_for_mixing->Mute();
     } else {
       // Only call the limiter if we have something to mix.
-      LimitMixedAudio(mixedAudio);
+      LimitMixedAudio(audio_frame_for_mixing);
     }
   }
 
-  {
-    CriticalSectionScoped cs(_cbCrit.get());
-    if (_mixReceiver != NULL) {
-      const AudioFrame** dummy = NULL;
-      _mixReceiver->NewMixedAudio(_id, *mixedAudio, dummy, 0);
-    }
-  }
-
-  // Reclaim all outstanding memory.
-  _audioFramePool->PushMemory(mixedAudio);
   ClearAudioFrameList(&mixList);
   ClearAudioFrameList(&rampOutList);
   ClearAudioFrameList(&additionalFramesList);
   {
     CriticalSectionScoped cs(_crit.get());
     _processCalls--;
   }
   return;
 }
 
-int32_t NewAudioConferenceMixerImpl::RegisterMixedStreamCallback(
-    OldAudioMixerOutputReceiver* mixReceiver) {
-  CriticalSectionScoped cs(_cbCrit.get());
-  if (_mixReceiver != NULL) {
-    return -1;
-  }
-  _mixReceiver = mixReceiver;
-  return 0;
-}
-
-int32_t NewAudioConferenceMixerImpl::UnRegisterMixedStreamCallback() {
-  CriticalSectionScoped cs(_cbCrit.get());
-  if (_mixReceiver == NULL) {
-    return -1;
-  }
-  _mixReceiver = NULL;
-  return 0;
-}
-
 int32_t NewAudioConferenceMixerImpl::SetOutputFrequency(
     const Frequency& frequency) {
   CriticalSectionScoped cs(_crit.get());
 
   _outputFrequency = frequency;
   _sampleSize =
       static_cast<size_t>((_outputFrequency * kProcessPeriodicityInMs) / 1000);
 
   return 0;
 }
@@ skipping 24 matching lines @@
     }
     bool success = false;
     if (mixable) {
       success = AddParticipantToList(participant, &_participantList);
     } else {
       success = RemoveParticipantFromList(participant, &_participantList);
     }
     if (!success) {
       WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                    "failed to %s participant", mixable ? "add" : "remove");
-      assert(false);
+      RTC_DCHECK(false);

[tommi, 2016/07/08 12:24:17]

RTC_NOTREACHED() here as well (and below).  Down the line, we should consider
deleting blocks of code like this (condition + body) and instead do:

RTC_CHECK(success);
or
RTC_DCHECK(success);

That is, if the above calls should really never fail.

[aleloi, 2016/07/08 12:57:40]

Done.

       return -1;
     }
 
     size_t numMixedNonAnonymous = _participantList.size();
     if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants) {
       numMixedNonAnonymous = kMaximumAmountOfMixedParticipants;
     }
     numMixedParticipants =
         numMixedNonAnonymous + _additionalParticipantList.size();
   }
@@ skipping 15 matching lines @@
     MixerAudioSource* participant,
     bool anonymous) {
   CriticalSectionScoped cs(_cbCrit.get());
   if (IsParticipantInList(*participant, _additionalParticipantList)) {
     if (anonymous) {
       return 0;
     }
     if (!RemoveParticipantFromList(participant, &_additionalParticipantList)) {
       WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                    "unable to remove participant from anonymous list");
-      assert(false);
+      RTC_DCHECK(false);
       return -1;
     }
     return AddParticipantToList(participant, &_participantList) ? 0 : -1;
   }
   if (!anonymous) {
     return 0;
   }
   const bool mixable =
       RemoveParticipantFromList(participant, &_participantList);
   if (!mixable) {
@@ skipping 23 matching lines @@
     freq = kSwbInHz;
   }
 
   if ((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) ||
       (freq == kLowestPossible)) {
     _minimumMixingFreq = freq;
     return 0;
   } else {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "SetMinimumMixingFrequency incorrect frequency: %i", freq);
-    assert(false);
+    RTC_DCHECK(false);
     return -1;
   }
 }
 
 // Check all AudioFrames that are to be mixed. The highest sampling frequency
 // found is the lowest that can be used without losing information.
 int32_t NewAudioConferenceMixerImpl::GetLowestMixingFrequency() const {
   const int participantListFrequency =
       GetLowestMixingFrequencyFromList(_participantList);
   const int anonymousListFrequency =
@@ skipping 46 matching lines @@
         (*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);
+      RTC_DCHECK(false);
       return;
     }
     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
     // assert until the problem is fixed on a higher level.
-    // assert(audioFrame->vad_activity_ != AudioFrame::kVadUnknown);
+    // RTC_DCHECK_NE(audioFrame->vad_activity_, AudioFrame::kVadUnknown);
     if (audioFrame->vad_activity_ == AudioFrame::kVadUnknown) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
                    "invalid VAD state from participant");
     }
 
     if (audioFrame->vad_activity_ == AudioFrame::kVadActive) {
       if (!wasMixed && !muted) {
         RampIn(*audioFrame);
       }
 
@@ skipping 17 matching lines @@
           RTC_DCHECK(!muted);  // Cannot replace with a muted frame.
           FrameAndMuteInfo replaceFrame = *replaceItem;
 
           bool replaceWasMixed = false;
           std::map<int, MixerAudioSource*>::const_iterator it =
               mixParticipantList->find(replaceFrame.frame->id_);
 
           // When a frame is pushed to |activeList| it is also pushed
           // to mixParticipantList with the frame's id. This means
           // that the Find call above should never fail.
-          assert(it != mixParticipantList->end());
+          RTC_DCHECK(it != mixParticipantList->end());
           replaceWasMixed = it->second->_mixHistory->WasMixed();
 
           mixParticipantList->erase(replaceFrame.frame->id_);
           activeList.erase(replaceItem);
 
           activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
           (*mixParticipantList)[audioFrame->id_] = *participant;
-          assert(mixParticipantList->size() <=
-                 kMaximumAmountOfMixedParticipants);
+          RTC_DCHECK_LE(mixParticipantList->size(),
+                        static_cast<size_t>(kMaximumAmountOfMixedParticipants));
 
           if (replaceWasMixed) {
             if (!replaceFrame.muted) {
               RampOut(*replaceFrame.frame);
             }
             rampOutList->push_back(replaceFrame);
-            assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
+            RTC_DCHECK_LE(
+                rampOutList->size(),
+                static_cast<size_t>(kMaximumAmountOfMixedParticipants));
           } else {
             _audioFramePool->PushMemory(replaceFrame.frame);
           }
         } else {
           if (wasMixed) {
             if (!muted) {
               RampOut(*audioFrame);
             }
             rampOutList->push_back(FrameAndMuteInfo(audioFrame, muted));
-            assert(rampOutList->size() <= kMaximumAmountOfMixedParticipants);
+            RTC_DCHECK_LE(
+                rampOutList->size(),
+                static_cast<size_t>(kMaximumAmountOfMixedParticipants));
           } else {
             _audioFramePool->PushMemory(audioFrame);
           }
         }
       } else {
         activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
         (*mixParticipantList)[audioFrame->id_] = *participant;
-        assert(mixParticipantList->size() <= kMaximumAmountOfMixedParticipants);
+        RTC_DCHECK_LE(mixParticipantList->size(),
+                      static_cast<size_t>(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);
+  RTC_DCHECK_LE(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);
+      RTC_DCHECK_LE(mixParticipantList->size(),
+                    static_cast<size_t>(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);
+      RTC_DCHECK_LE(mixParticipantList->size(),
+                    static_cast<size_t>(kMaximumAmountOfMixedParticipants));
     } else {
       _audioFramePool->PushMemory((*iter)->audioFrame);
     }
     delete *iter;
   }
-  assert(*maxAudioFrameCounter + mixListStartSize >= mixList->size());
+  RTC_DCHECK_GE(*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);
+      RTC_DCHECK(false);
       return;
     }
     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,
                "UpdateMixedStatus(mixedParticipantsMap)");
-  assert(mixedParticipantsMap.size() <= kMaximumAmountOfMixedParticipants);
+  RTC_DCHECK_LE(mixedParticipantsMap.size(),
+                static_cast<size_t>(kMaximumAmountOfMixedParticipants));
 
   // Loop through all participants. If they are in the mix map they
   // were mixed.
   for (MixerAudioSourceList::const_iterator participant =
            _participantList.begin();
        participant != _participantList.end(); ++participant) {
     bool isMixed = false;
     for (std::map<int, MixerAudioSource*>::const_iterator it =
              mixedParticipantsMap.begin();
          it != mixedParticipantsMap.end(); ++it) {
@@ skipping 54 matching lines @@
       // Participant is no longer mixed, reset to default.
       participant->_mixHistory->ResetMixedStatus();
       return true;
     }
   }
   return false;
 }
 
 int32_t NewAudioConferenceMixerImpl::MixFromList(
     AudioFrame* mixedAudio,
-    const AudioFrameList& audioFrameList) const {
-  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
+    const AudioFrameList& audioFrameList,
+    int32_t id,
+    bool use_limiter) {
+  WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id,
                "MixFromList(mixedAudio, audioFrameList)");
   if (audioFrameList.empty())
     return 0;
 
   uint32_t position = 0;
 
-  if (_numMixedParticipants == 1) {
+  if (audioFrameList.size() == 1) {
     mixedAudio->timestamp_ = audioFrameList.front().frame->timestamp_;
     mixedAudio->elapsed_time_ms_ =
         audioFrameList.front().frame->elapsed_time_ms_;
   } else {
     // TODO(wu): Issue 3390.
     // Audio frame timestamp is only supported in one channel case.
     mixedAudio->timestamp_ = 0;
     mixedAudio->elapsed_time_ms_ = -1;
   }
 
   for (AudioFrameList::const_iterator iter = audioFrameList.begin();
        iter != audioFrameList.end(); ++iter) {
     if (position >= kMaximumAmountOfMixedParticipants) {
       WEBRTC_TRACE(
-          kTraceMemory, kTraceAudioMixerServer, _id,
+          kTraceMemory, kTraceAudioMixerServer, id,
           "Trying to mix more than max amount of mixed participants:%d!",
           kMaximumAmountOfMixedParticipants);
       // Assert and avoid crash
-      assert(false);
+      RTC_DCHECK(false);
       position = 0;
     }
     if (!iter->muted) {
-      MixFrames(mixedAudio, iter->frame, use_limiter_);
+      MixFrames(mixedAudio, iter->frame, use_limiter);
     }
 
     position++;
   }
 
   return 0;
 }
 
 // TODO(andrew): consolidate this function with MixFromList.
 int32_t NewAudioConferenceMixerImpl::MixAnonomouslyFromList(
@@ skipping 31 matching lines @@
   // is enabled (moving from one to two participants) it has the potential to
   // create discontinuities in the mixed frame.
   //
   // Instead we double the frame (with addition since left-shifting a
   // negative value is undefined).
   *mixedAudio += *mixedAudio;
 
   if (error != _limiter->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "Error from AudioProcessing: %d", error);
-    assert(false);
+    RTC_DCHECK(false);
     return false;
   }
   return true;
 }
 }  // namespace webrtc