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

talk/app/webrtc/mediastreamhandler.cc

 /*
  * 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 50 matching lines @@
 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
   rtc::CritScope lock(&lock_);
   if (sink_)
     sink_->OnClose();
 }
 
 void LocalAudioSinkAdapter::OnData(const void* audio_data,
                                    int bits_per_sample,
                                    int sample_rate,
                                    int number_of_channels,
-                                   int number_of_frames) {
+                                   size_t number_of_frames) {
   rtc::CritScope lock(&lock_);
   if (sink_) {
     sink_->OnData(audio_data, bits_per_sample, sample_rate,
                   number_of_channels, number_of_frames);
   }
 }
 
 void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
   rtc::CritScope lock(&lock_);
   ASSERT(!sink || !sink_);
@@ skipping 395 matching lines @@
     if ((*it)->stream() == stream) {
       (*it)->Stop();
       delete *it;
       streamhandlers->erase(it);
       break;
     }
   }
 }
 
 }  // namespace webrtc