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

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 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 1551 matching lines @@
     success = false;
   }
   return success;
 }
 
 // Implementing method from WebRtc VoEMediaProcess interface
 // Do not lock mux_channel_cs_ in this callback.
 void WebRtcVoiceEngine::Process(int channel,
                                 webrtc::ProcessingTypes type,
                                 int16_t audio10ms[],
-                                int length,
+                                size_t length,
                                 int sampling_freq,
                                 bool is_stereo) {
     rtc::CritScope cs(&signal_media_critical_);
     AudioFrame frame(audio10ms, length, sampling_freq, is_stereo);
     if (type == webrtc::kPlaybackAllChannelsMixed) {
       SignalRxMediaFrame(rx_processor_ssrc_, MPD_RX, &frame);
     } else if (type == webrtc::kRecordingPerChannel) {
       SignalTxMediaFrame(tx_processor_ssrc_, MPD_TX, &frame);
     } else {
       LOG(LS_WARNING) << "Media Processing invoked unexpectedly."
@@ skipping 75 matching lines @@
     renderer_->SetSink(NULL);
     renderer_ = NULL;
   }
 
   // AudioRenderer::Sink implementation.
   // This method is called on the audio thread.
   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 {
     voe_audio_transport_->OnData(channel_,
                                  audio_data,
                                  bits_per_sample,
                                  sample_rate,
                                  number_of_channels,
                                  number_of_frames);
   }
 
   // Callback from the |renderer_| when it is going away. In case Start() has
   // never been called, this callback won't be triggered.
@@ skipping 2006 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE