Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

Index: webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
index 4b36fffed686b3154ebdd1c53ee7e9354b31150e..e80ad3f6dae3f3ddf90309725f7c01ac6b42a5a2 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
@@ -25,7 +25,7 @@
 void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
 
 int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
-                                 int32_t packet_size,
+                                 size_t packet_size,
                                  uint16_t rtp_seq_number,
                                  uint32_t send_ts, uint32_t arr_ts,
                                  enum IsacSamplingRate encoderSampRate,
@@ -228,6 +228,6 @@ void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
 
 void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
 
-void WebRtcIsac_AutoCorr(double* r, const double* x, int N, int order);
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */