b8b6fbb7a5d2f5a14f7f6f81c253747aa28e4c7f |
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10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint build/include errors fixed in rtp_rtcp module BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1505993003 Cr-Commit-Position: refs/heads/master@{#10971}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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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/rtcp_receiver_help.h
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6b8d3551681f40b880507cecc88f478a12383cc7 |
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24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation." Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc BUG=webrtc:4173 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1362303002 . Cr-Commit-Position: refs/heads/master@{#10052}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
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23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ ) Reason for revert: Breaking some Android bots. https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29 Original issue's description: > Wire up send-side bandwidth estimation. > > BUG=webrtc:4173 > > Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547 > Cr-Commit-Position: refs/heads/master@{#10012} TBR=stefan@webrtc.org, kjellander@webrtc.org NOPRESUBMIT=false NOTREECHECKS=false NOTRY=false BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1362923002 . Cr-Commit-Position: refs/heads/master@{#10029}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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ef165eefc79cf28bb67779afe303cc2365885547 |
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22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1338203003 Cr-Commit-Position: refs/heads/master@{#10012}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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3c089d751ede283e21e186885eaf705c3257ccd2 |
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16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros. We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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00b8f6b3643332cce1ee711715f7fbb824d793ca |
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26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
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12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values. Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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88fbb2d86b33a3886bba1af4d098efa2c19eb1e7 |
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21-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. Same as https://webrtc-codereview.appspot.com/19519004. The issue in http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux... is solved by this change http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing... (tested locally). BUG=3380 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17619005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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2fa7f79094bfa283e0ff2b086be511e65c24c69e |
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21-May-2014 |
mcasas@webrtc.org <mcasas@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6202 "Switch to using base/constructormagic.h and remove ..." > Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. > > BUG=N/A > R=andrew@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/19519004 TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14579007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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125ffd709dee39214e54d80fb277da66adc9ebd3 |
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20-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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8469f7b328ec980f80fa79931b4e07872d0feb23 |
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02-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added support for sending and receiving RTCP XR packets: - Receiver reference time report block - DLRR report block (RFC3611). BUG=1613 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2196010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4898 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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28a331eededf17dc3a0860bb1bdf5b2dc3f9e763 |
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17-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support for multiple report blocks. Use a weighted average of fraction loss for bandwidth estimation. TEST=trybots and vie_auto_test --automated BUG=1811 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2198004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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d900e8bea84c474696bf0219aed1353ce65ffd8e |
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03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments. BUG= R=mflodman@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1760006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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2f44673d665899ca788ae44247a9a7f4764f5e2b |
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08-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t for rtp_rtcp/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1279007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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becf9c897c41eea3f021f99d87889c32c78b0de9 |
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01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix mismatch between different NACK list lengths and packet buffers. This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
|