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

webrtc/common_types.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 363 matching lines @@
     int32_t clockDriftPPM;
     // average packet waiting time in the jitter buffer (ms)
     int meanWaitingTimeMs;
     // median packet waiting time in the jitter buffer (ms)
     int medianWaitingTimeMs;
     // min packet waiting time in the jitter buffer (ms)
     int minWaitingTimeMs;
     // max packet waiting time in the jitter buffer (ms)
     int maxWaitingTimeMs;
     // added samples in off mode due to packet loss
-    int addedSamples;
+    size_t addedSamples;
 };
 
 // Statistics for calls to AudioCodingModule::PlayoutData10Ms().
 struct AudioDecodingCallStats {
   AudioDecodingCallStats()
       : calls_to_silence_generator(0),
         calls_to_neteq(0),
         decoded_normal(0),
         decoded_plc(0),
         decoded_cng(0),
@@ skipping 505 matching lines @@
 class StreamDataCountersCallback {
  public:
   virtual ~StreamDataCountersCallback() {}
 
   virtual void DataCountersUpdated(const StreamDataCounters& counters,
                                    uint32_t ssrc) = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_