ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.h
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d42a3adf429ad27779bea1789f53b76e52388583 |
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07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove partially defined WebRtcRTPHeader from Parse(). It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change. To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28919004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.h
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730d2707713c4240070af17e56edd10d039bafd2 |
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29-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove callback from RtpDepacketizer::Parse(). BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30489004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.h
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b5e6bfc76a32a588da2400636688d34a71a2f47d |
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12-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make RTPSender/RTPReceiver generic. Changes include, 1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric. 2) Introduce class RtpDepacketizerVp8. 3) Make RTPSenderVideo::SendH264 generic and used by all packetizers. 4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to RtpPacketizer/RtpDePacketizer sub-classes. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26399004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.h
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2ec560606be6519dc4e32a1e6855b0f362ca498d |
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31-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.h
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