Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index fe6fd87dfd791b53247bd26cc59fbd73a08f3187..0a40316b713025028fd67be64f9d75a08d124168 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -90,11 +90,11 @@ class AudioEncoder {
   // the encoder may vary the number of 10 ms frames from packet to packet, but
   // it must decide the length of the next packet no later than when outputting
   // the preceding packet.
-  virtual int Num10MsFramesInNextPacket() const = 0;
+  virtual size_t Num10MsFramesInNextPacket() const = 0;
 
   // Returns the maximum value that can be returned by
   // Num10MsFramesInNextPacket().
-  virtual int Max10MsFramesInAPacket() const = 0;
+  virtual size_t Max10MsFramesInAPacket() const = 0;
 
   // Returns the current target bitrate in bits/s. The value -1 means that the
   // codec adapts the target automatically, and a current target cannot be