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Use size_t more consistently for packet/payload lengths.

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -40,7 +40,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Called when we receive an RTCP packet.
virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
uint16_t incoming_packet_length) OVERRIDE;
size_t incoming_packet_length) OVERRIDE;
virtual void SetRemoteSSRC(const uint32_t ssrc) OVERRIDE;
@ -120,7 +120,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
const uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
const uint32_t payload_size,
const size_t payload_size,
const RTPFragmentationHeader* fragmentation = NULL,
const RTPVideoHeader* rtp_video_hdr = NULL) OVERRIDE;
@ -130,7 +130,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
bool retransmission) OVERRIDE;
// Returns the number of padding bytes actually sent, which can be more or
// less than |bytes|.
virtual int TimeToSendPadding(int bytes) OVERRIDE;
virtual size_t TimeToSendPadding(size_t bytes) OVERRIDE;
virtual bool GetSendSideDelay(int* avg_send_delay_ms,
int* max_send_delay_ms) const OVERRIDE;
@ -179,7 +179,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual int32_t ResetSendDataCountersRTP() OVERRIDE;
// Statistics of the amount of data sent and received.
virtual int32_t DataCountersRTP(uint32_t* bytes_sent,
virtual int32_t DataCountersRTP(size_t* bytes_sent,
uint32_t* packets_sent) const OVERRIDE;
// Get received RTCP report, sender info.