Convert channel counts to size_t.

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 32 matching lines @@
     // We only support mono-to-stereo.
     assert(mixed_frame->num_channels_ == 2 &&
            frame->num_channels_ == 1);
     AudioFrameOperations::MonoToStereo(frame);
   }
 
   *mixed_frame += *frame;
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
-int MaxNumChannels(const AudioFrameList* list) {
-  int max_num_channels = 1;
+size_t MaxNumChannels(const AudioFrameList* list) {
+  size_t max_num_channels = 1;
   for (AudioFrameList::const_iterator iter = list->begin();
        iter != list->end();
        ++iter) {
     max_num_channels = std::max(max_num_channels, (*iter)->num_channels_);
   }
   return max_num_channels;
 }
 
 }  // namespace
 
@@ skipping 206 matching lines @@
     }
 
     int retval = 0;
     {
         CriticalSectionScoped cs(_crit.get());
 
         // 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 int num_mixed_channels = std::max(MaxNumChannels(&mixList),
+        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
@@ skipping 627 matching lines @@
 
     if(error != _limiter->kNoError) {
         WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                      "Error from AudioProcessing: %d", error);
         assert(false);
         return false;
     }
     return true;
 }
 }  // namespace webrtc