Updated mixer unittests and fixed a related bug in the new mixer.

webrtc/modules/audio_mixer/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 169 matching lines @@
   if (limiter_->gain_control()->Enable(true) != limiter_->kNoError)
     return false;
 
   return true;
 }
 
 void NewAudioConferenceMixerImpl::Mix(int sample_rate,
                                       size_t number_of_channels,
                                       AudioFrame* audio_frame_for_mixing) {
   RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);
-  size_t remainingAudioSourcesAllowedToMix = kMaximumAmountOfMixedAudioSources;
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   AudioFrameList mixList;
   AudioFrameList additionalFramesList;
   std::map<int, MixerAudioSource*> mixedAudioSourcesMap;
   {
     CriticalSectionScoped cs(cb_crit_.get());
     Frequency mixing_frequency;
 
     switch (sample_rate) {
       case 8000:
@@ skipping 10 matching lines @@
         break;
       default:
         RTC_NOTREACHED();
         return;
     }
 
     if (OutputFrequency() != mixing_frequency) {
       SetOutputFrequency(mixing_frequency);
     }
 
-    mixList = UpdateToMix(remainingAudioSourcesAllowedToMix);
-    remainingAudioSourcesAllowedToMix -= mixList.size();

[aleloi, 2016/08/18 09:21:14]

The changes above are a simplification of the handling for the number of
participants. With the new mixer, it may happen that mixList.size() >
kMaximumAmountOfMixedAudioSources, because RampOut participants are also placed
in mixList.

+    mixList = UpdateToMix(kMaximumAmountOfMixedAudioSources);
     GetAdditionalAudio(&additionalFramesList);
   }
 
   for (FrameAndMuteInfo& frame_and_mute : mixList) {
     RemixFrame(frame_and_mute.frame, number_of_channels);
   }
   for (FrameAndMuteInfo& frame_and_mute : additionalFramesList) {
     RemixFrame(frame_and_mute.frame, number_of_channels);
   }
 
@@ skipping 286 matching lines @@
         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 >= kMaximumAmountOfMixedAudioSources) {
-      WEBRTC_TRACE(
-          kTraceMemory, kTraceAudioMixerServer, id,
-          "Trying to mix more than max amount of mixed audio sources:%d!",
-          kMaximumAmountOfMixedAudioSources);
-      // Assert and avoid crash
-      RTC_NOTREACHED();
-      position = 0;
-    }

[aleloi, 2016/08/18 09:21:14]

It may happen that audioFrameList.size() > kMax..Sources, because RampOut
participants are placed in mixList.

     if (!iter->muted) {
       MixFrames(mixedAudio, iter->frame, use_limiter);
     }
 
     position++;
   }
 
   return 0;
 }
 
@@ skipping 55 matching lines @@
 }
 
 uint32_t NewAudioConferenceMixerImpl::GetOutputAudioLevelFullRange() {
   int16_t current_level = audio_level_.LevelFullRange();
   uint32_t level = static_cast<uint32_t>(current_level);
   WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
                "GetAudioOutputLevelFullRange() => level=%u", level);
   return level;
 }
 }  // namespace webrtc