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
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index f6b5553..4ee4fa2 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -30,9 +30,11 @@
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
~TestPacketization();
virtual int32_t SendData(
- const FrameType frameType, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+ const FrameType frameType,
+ const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const uint8_t* payloadData,
+ const size_t payloadSize,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
private:
@@ -92,8 +94,8 @@
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
- uint16_t _realPayloadSizeBytes;
- uint16_t _payloadSizeBytes;
+ size_t _realPayloadSizeBytes;
+ size_t _payloadSizeBytes;
uint32_t _nextTime;
};