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

webrtc/modules/audio_coding/neteq/decision_logic_fax.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 namespace webrtc {
 
 Operations DecisionLogicFax::GetDecisionSpecialized(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
     bool* reset_decoder) {
   assert(playout_mode_ == kPlayoutFax || playout_mode_ == kPlayoutOff);
   uint32_t target_timestamp = sync_buffer.end_timestamp();
   uint32_t available_timestamp = 0;
   int is_cng_packet = 0;
   if (packet_header) {
     available_timestamp = packet_header->timestamp;
@@ skipping 57 matching lines @@
             assert(0);
             return kUndefined;
         }
       }
     }
   }
 }
 
 
 }  // namespace webrtc