Use size_t more consistently for packet/payload lengths.

See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/video_coding/main/interface/video_coding.h

@@ -467,8 +467,8 @@ public:
     // Return value      : VCM_OK, on success.
     //                     < 0,         on error.
     virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
-                                       uint32_t payloadLength,
-                                       const WebRtcRTPHeader& rtpInfo) = 0;
+                                   size_t payloadLength,
+                                   const WebRtcRTPHeader& rtpInfo) = 0;
 
     // Minimum playout delay (Used for lip-sync). This is the minimum delay required
     // to sync with audio. Not included in  VideoCodingModule::Delay()

webrtc/modules/video_coding/main/interface/video_coding_defines.h

@@ -75,7 +75,7 @@ class VCMPacketizationCallback {
       uint32_t timeStamp,
       int64_t capture_time_ms,
       const uint8_t* payloadData,
-      uint32_t payloadSize,
+      size_t payloadSize,
       const RTPFragmentationHeader& fragmentationHeader,
       const RTPVideoHeader* rtpVideoHdr) = 0;
  protected: