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

talk/app/webrtc/mediastreaminterface.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 169 matching lines @@
   virtual void UnregisterAudioObserver(AudioObserver* observer) {}
 };
 
 // Interface for receiving audio data from a AudioTrack.
 class AudioTrackSinkInterface {
  public:
   virtual void OnData(const void* audio_data,
                       int bits_per_sample,
                       int sample_rate,
                       int number_of_channels,
-                      int number_of_frames) = 0;
+                      size_t number_of_frames) = 0;
  protected:
   virtual ~AudioTrackSinkInterface() {}
 };
 
 // Interface of the audio processor used by the audio track to collect
 // statistics.
 class AudioProcessorInterface : public rtc::RefCountInterface {
  public:
   struct AudioProcessorStats {
     AudioProcessorStats() : typing_noise_detected(false),
@@ skipping 72 matching lines @@
   virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
   virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
 
  protected:
   virtual ~MediaStreamInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_