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/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index cdb99c0..4ab32b9 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -27,7 +27,7 @@
// TODO(turajs): Write constructor for this structure.
struct ACMTestFrameSizeStats {
uint16_t frameSizeSample;
- int16_t maxPayloadLen;
+ size_t maxPayloadLen;
uint32_t numPackets;
uint64_t totalPayloadLenByte;
uint64_t totalEncodedSamples;
@@ -39,7 +39,7 @@
struct ACMTestPayloadStats {
bool newPacket;
int16_t payloadType;
- int16_t lastPayloadLenByte;
+ size_t lastPayloadLenByte;
uint32_t lastTimestamp;
ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
};
@@ -51,9 +51,11 @@
~Channel();
virtual int32_t SendData(
- const FrameType frameType, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
- const uint16_t payloadSize,
+ FrameType frameType,
+ uint8_t payloadType,
+ uint32_t timeStamp,
+ const uint8_t* payloadData,
+ size_t payloadSize,
const RTPFragmentationHeader* fragmentation) OVERRIDE;
void RegisterReceiverACM(AudioCodingModule *acm);
@@ -93,7 +95,7 @@
}
private:
- void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
+ void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
AudioCodingModule* _receiverACM;
uint16_t _seqNo;