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/utility/source/coder.cc

@@ -54,7 +54,7 @@ int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
 int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
                            uint32_t sampFreqHz,
                            const int8_t*  incomingPayload,
-                           int32_t  payloadLength)
+                           size_t  payloadLength)
 {
     if (payloadLength > 0)
     {
@@ -79,7 +79,7 @@ int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
 
 int32_t AudioCoder::Encode(const AudioFrame& audio,
                            int8_t* encodedData,
-                           uint32_t& encodedLengthInBytes)
+                           size_t& encodedLengthInBytes)
 {
     // Fake a timestamp in case audio doesn't contain a correct timestamp.
     // Make a local copy of the audio frame since audio is const
@@ -109,7 +109,7 @@ int32_t AudioCoder::SendData(
     uint8_t   /* payloadType */,
     uint32_t  /* timeStamp */,
     const uint8_t*  payloadData,
-    uint16_t  payloadSize,
+    size_t  payloadSize,
     const RTPFragmentationHeader* /* fragmentation*/)
 {
     memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);