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Use size_t more consistently for packet/payload lengths.

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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
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
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14
webrtc/modules/audio_coding/main/test/Channel.h
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@ -27,7 +27,7 @@ class CriticalSectionWrapper;
// 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 ACMTestFrameSizeStats {
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 @@ class Channel : public AudioPacketizationCallback {
~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 @@ class Channel : public AudioPacketizationCallback {
}
private:
void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
AudioCodingModule* _receiverACM;
uint16_t _seqNo;