Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/decision_logic_normal.cc

Index: webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index e985ee0aa394c274ffda9ffc5bcf21429ac676de..bd05d8f0472d53880908d57d21a6224ef3e72d51 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -27,7 +27,7 @@ namespace webrtc {
 Operations DecisionLogicNormal::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,
@@ -149,7 +149,7 @@ Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
 Operations DecisionLogicNormal::FuturePacketAvailable(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
-    int decoder_frame_length,
+    size_t decoder_frame_length,
     Modes prev_mode,
     uint32_t target_timestamp,
     uint32_t available_timestamp,
@@ -172,9 +172,9 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
     }
   }
 
-  const int samples_left = static_cast<int>(sync_buffer.FutureLength() -
-      expand.overlap_length());
-  const int cur_size_samples = samples_left +
+  const size_t samples_left =
+      sync_buffer.FutureLength() - expand.overlap_length();
+  const size_t cur_size_samples = samples_left +
       packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
 
   // If previous was comfort noise, then no merge is needed.