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

webrtc/voice_engine/transmit_mixer.h

Index: webrtc/voice_engine/transmit_mixer.h
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 919de131238f5123ddb9fb4c16d7a9ca5423162b..8bbb421a3b55216ac6d892404193e23bb2bc3d93 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -51,7 +51,7 @@ public:
         AudioProcessing* audioProcessingModule);
 
     int32_t PrepareDemux(const void* audioSamples,
-                         uint32_t nSamples,
+                         size_t nSamples,
                          uint8_t  nChannels,
                          uint32_t samplesPerSec,
                          uint16_t totalDelayMS,
@@ -173,7 +173,7 @@ private:
     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);