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

talk/app/webrtc/mediastreamhandler.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 64 matching lines @@
  public:
   LocalAudioSinkAdapter();
   virtual ~LocalAudioSinkAdapter();
 
  private:
   // AudioSinkInterface implementation.
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
               int number_of_channels,
-              int number_of_frames) override;
+              size_t number_of_frames) override;
 
   // cricket::AudioRenderer implementation.
   void SetSink(cricket::AudioRenderer::Sink* sink) override;
 
   cricket::AudioRenderer::Sink* sink_;
   // Critical section protecting |sink_|.
   rtc::CriticalSection lock_;
 };
 
 // LocalAudioTrackHandler listen to events on a local AudioTrack instance
@@ skipping 188 matching lines @@
 
   StreamHandlerList local_streams_handlers_;
   StreamHandlerList remote_streams_handlers_;
   AudioProviderInterface* audio_provider_;
   VideoProviderInterface* video_provider_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMHANDLER_H_