Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/main/acm2/codec_manager.cc

Index: webrtc/modules/audio_coding/main/acm2/codec_manager.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
index cad6ee908939b9c1523924f5b2d1b372b7077f1d..7b9c7ed5a949db37b8dbd842a1389fd99efbf479 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
@@ -326,10 +326,10 @@ void CodecManager::RegisterEncoder(
   // Make up a CodecInst.
   send_codec_inst_.channels = external_speech_encoder->NumChannels();
   send_codec_inst_.plfreq = external_speech_encoder->SampleRateHz();
-  send_codec_inst_.pacsize =
-      rtc::CheckedDivExact(external_speech_encoder->Max10MsFramesInAPacket() *
-                               send_codec_inst_.plfreq,
-                           100);
+  send_codec_inst_.pacsize = rtc::CheckedDivExact(
+      static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
+                       send_codec_inst_.plfreq),
+      100);
   send_codec_inst_.pltype = -1;  // Not valid.
   send_codec_inst_.rate = -1;    // Not valid.
   static const char kName[] = "external";