Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_processing/aec/aec_resampler.h

Index: webrtc/modules/audio_processing/aec/aec_resampler.h
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.h b/webrtc/modules/audio_processing/aec/aec_resampler.h
index a37499258f317217a94806658aac5fc3ce5af5da..a5002c155a4a5fff6f338223f01aef4dc2aa8593 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.h
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.h
@@ -31,9 +31,9 @@ int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
 // Resamples input using linear interpolation.
 void WebRtcAec_ResampleLinear(void* resampInst,
                               const float* inspeech,
-                              int size,
+                              size_t size,
                               float skew,
                               float* outspeech,
-                              int* size_out);
+                              size_t* size_out);
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_