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

webrtc/modules/audio_coding/neteq/decision_logic_fax.h

Index: webrtc/modules/audio_coding/neteq/decision_logic_fax.h
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_fax.h b/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
index d9f8db9652ee62dff831c5498e5fd3a4dbca871f..861e2fa87d3d7e6ce96734f3daa644597658b975 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
@@ -23,7 +23,7 @@ class DecisionLogicFax : public DecisionLogic {
  public:
   // Constructor.
   DecisionLogicFax(int fs_hz,
-                   int output_size_samples,
+                   size_t output_size_samples,
                    NetEqPlayoutMode playout_mode,
                    DecoderDatabase* decoder_database,
                    const PacketBuffer& packet_buffer,
@@ -46,7 +46,7 @@ class DecisionLogicFax : public DecisionLogic {
   // remain true if it was true before the call).
   Operations 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,