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

webrtc/voice_engine/transmit_mixer.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 33 matching lines @@
     static void Destroy(TransmitMixer*& mixer);
 
     int32_t SetEngineInformation(ProcessThread& processThread,
                                  Statistics& engineStatistics,
                                  ChannelManager& channelManager);
 
     int32_t SetAudioProcessingModule(
         AudioProcessing* audioProcessingModule);
 
     int32_t PrepareDemux(const void* audioSamples,
-                         uint32_t nSamples,
+                         size_t nSamples,
                          uint8_t  nChannels,
                          uint32_t samplesPerSec,
                          uint16_t totalDelayMS,
                          int32_t  clockDrift,
                          uint16_t currentMicLevel,
                          bool keyPressed);
 
 
     int32_t DemuxAndMix();
     // Used by the Chrome to pass the recording data to the specific VoE
@@ skipping 101 matching lines @@
   bool IsStereoChannelSwappingEnabled();
 
 private:
     TransmitMixer(uint32_t instanceId);
 
     // Gets the maximum sample rate and number of channels over all currently
     // sending codecs.
     void GetSendCodecInfo(int* max_sample_rate, int* max_channels);
 
     void GenerateAudioFrame(const int16_t audioSamples[],
-                            int nSamples,
+                            size_t nSamples,
                             int nChannels,
                             int samplesPerSec);
     int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
     int32_t MixOrReplaceAudioWithFile(
         int mixingFrequency);
 
     void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
                       bool key_pressed);
 
@@ skipping 43 matching lines @@
     bool stereo_codec_;
     bool swap_stereo_channels_;
     rtc::scoped_ptr<int16_t[]> mono_buffer_;
 };
 
 }  // namespace voe
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H