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

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * 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 112 matching lines @@
                          VoiceProcessor* voice_processor,
                          MediaProcessorDirection direction);
   bool UnregisterProcessor(uint32 ssrc,
                            VoiceProcessor* voice_processor,
                            MediaProcessorDirection direction);
 
   // Method from webrtc::VoEMediaProcess
   void Process(int channel,
                webrtc::ProcessingTypes type,
                int16_t audio10ms[],
-               int length,
+               size_t length,
                int sampling_freq,
                bool is_stereo) override;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel *channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
 
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
@@ skipping 310 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_