Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_conference_mixer/source/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 269 matching lines @@
 
         // Find the max channels over all mixing lists.
         const int 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 += _sampleSize;
+        _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 <= kAudioProcMaxNativeSampleRateHz;
 
         MixFromList(*mixedAudio, &mixList);
         MixAnonomouslyFromList(*mixedAudio, &additionalFramesList);
         MixAnonomouslyFromList(*mixedAudio, &rampOutList);
 
@@ skipping 49 matching lines @@
     }
     _mixReceiver = NULL;
     return 0;
 }
 
 int32_t AudioConferenceMixerImpl::SetOutputFrequency(
     const Frequency frequency) {
     CriticalSectionScoped cs(_crit.get());
 
     _outputFrequency = frequency;
-    _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;
+    _sampleSize =
+        static_cast<size_t>((_outputFrequency*kProcessPeriodicityInMs) / 1000);
 
     return 0;
 }
 
 AudioConferenceMixer::Frequency
 AudioConferenceMixerImpl::OutputFrequency() const {
     CriticalSectionScoped cs(_crit.get());
     return _outputFrequency;
 }
 
@@ skipping 551 matching lines @@
 
     if(error != _limiter->kNoError) {
         WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                      "Error from AudioProcessing: %d", error);
         assert(false);
         return false;
     }
     return true;
 }
 }  // namespace webrtc