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Some refactoring inside rtp_rtcp/.

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Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-07-08 12:10:51 +00:00
parent 241a9b0b65
commit 62bafae661
45 changed files with 384 additions and 400 deletions
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8
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
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@ -22,10 +22,10 @@
namespace webrtc {
using ModuleRTPUtility::GetCurrentRTP;
using ModuleRTPUtility::Payload;
using ModuleRTPUtility::RTPPayloadParser;
using ModuleRTPUtility::StringCompare;
using RtpUtility::GetCurrentRTP;
using RtpUtility::Payload;
using RtpUtility::RTPPayloadParser;
using RtpUtility::StringCompare;
RtpReceiver* RtpReceiver::CreateVideoReceiver(
int id, Clock* clock,