Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h b/webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h
index 553d8ad4b8cd5d5db1206540d058a4b1bf650d0a..c1184e16a8ddd74c144c222e08be575125ace191 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h
@@ -74,11 +74,11 @@ class AudioEncoderMutableImpl : public P {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->RtpTimestampRateHz();
   }
-  int Num10MsFramesInNextPacket() const override {
+  size_t Num10MsFramesInNextPacket() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->Num10MsFramesInNextPacket();
   }
-  int Max10MsFramesInAPacket() const override {
+  size_t Max10MsFramesInAPacket() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->Max10MsFramesInAPacket();
   }