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

webrtc/voice_engine/channel.h

 /*
  *  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 413 matching lines @@
     int8_t OutputEnergyLevel() const
     {
         return _outputAudioLevel.Level();
     }
     uint32_t Demultiplex(const AudioFrame& audioFrame);
     // Demultiplex the data to the channel's |_audioFrame|. The difference
     // between this method and the overloaded method above is that |audio_data|
     // does not go through transmit_mixer and APM.
     void Demultiplex(const int16_t* audio_data,
                      int sample_rate,
-                     int number_of_frames,
+                     size_t number_of_frames,
                      int number_of_channels);
     uint32_t PrepareEncodeAndSend(int mixingFrequency);
     uint32_t EncodeAndSend();
 
     // Associate to a send channel.
     // Used for obtaining RTT for a receive-only channel.
     void set_associate_send_channel(const ChannelOwner& channel) {
       assert(_channelId != channel.channel()->ChannelId());
       CriticalSectionScoped lock(assoc_send_channel_lock_.get());
       associate_send_channel_ = channel;
@@ skipping 133 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_