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

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 230 matching lines @@
   void DisableNack() override;
 
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
 
  private:
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
-    uint16_t length_per_channel;
+    size_t length_per_channel;
     uint8_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
   };
 
   // This member class writes values to the named UMA histogram, but only if
   // the value has changed since the last time (and always for the first call).
   class ChangeLogger {
    public:
@@ skipping 143 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_