fac0655fd7fe0b40ef50dc5b7f11ea44d72cec6c |
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25-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) Relanding after fixing CallAndModifyStream to account for new procedures for adding/removing a track from a stream. Original issue's description: > Adding the ability to create an RtpSender without a track. > > This CL also changes AddStream to immediately create a sender, rather > than waiting until the track is seen in SDP. And the PeerConnection now > builds the list of "send streams" from the list of senders, rather than > the collection of local media streams. > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > Cr-Commit-Position: refs/heads/master@{#10414} Review URL: https://codereview.webrtc.org/1468113002 Cr-Commit-Position: refs/heads/master@{#10790}
/external/webrtc/webrtc/base/helpers.cc
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5def7b9fdea0d027bca3df734d86fb877a83bdbf |
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20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) Reason for revert: Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection. Original issue's description: > Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) > > Reason for revert: > Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream. > > Original issue's description: > > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) > > > > Reason for revert: > > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. > > > > Original issue's description: > > > Adding the ability to create an RtpSender without a track. > > > > > > This CL also changes AddStream to immediately create a sender, rather > > > than waiting until the track is seen in SDP. And the PeerConnection now > > > builds the list of "send streams" from the list of senders, rather than > > > the collection of local media streams. > > > > > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > > > Cr-Commit-Position: refs/heads/master@{#10414} > > > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > > > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb > > Cr-Commit-Position: refs/heads/master@{#10417} > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae > Cr-Commit-Position: refs/heads/master@{#10730} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1460323002 Cr-Commit-Position: refs/heads/master@{#10732}
/external/webrtc/webrtc/base/helpers.cc
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6834fa10f142bf5e2275142acb834898911d09ae |
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20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) Reason for revert: Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream. Original issue's description: > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) > > Reason for revert: > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. > > Original issue's description: > > Adding the ability to create an RtpSender without a track. > > > > This CL also changes AddStream to immediately create a sender, rather > > than waiting until the track is seen in SDP. And the PeerConnection now > > builds the list of "send streams" from the list of senders, rather than > > the collection of local media streams. > > > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > > Cr-Commit-Position: refs/heads/master@{#10414} > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb > Cr-Commit-Position: refs/heads/master@{#10417} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1413983004 Cr-Commit-Position: refs/heads/master@{#10730}
/external/webrtc/webrtc/base/helpers.cc
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8f46c63f6f764254892f4111b54aa1cc8f32eeeb |
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26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) Reason for revert: Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. Original issue's description: > Adding the ability to create an RtpSender without a track. > > This CL also changes AddStream to immediately create a sender, rather > than waiting until the track is seen in SDP. And the PeerConnection now > builds the list of "send streams" from the list of senders, rather than > the collection of local media streams. > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > Cr-Commit-Position: refs/heads/master@{#10414} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1426443007 Cr-Commit-Position: refs/heads/master@{#10417}
/external/webrtc/webrtc/base/helpers.cc
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ac9d92ccbe2b29590c53f702e11dc625820480d5 |
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26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding the ability to create an RtpSender without a track. This CL also changes AddStream to immediately create a sender, rather than waiting until the track is seen in SDP. And the PeerConnection now builds the list of "send streams" from the list of senders, rather than the collection of local media streams. Review URL: https://codereview.webrtc.org/1413713003 Cr-Commit-Position: refs/heads/master@{#10414}
/external/webrtc/webrtc/base/helpers.cc
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0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
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07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types. Removes the use of uint8, etc. in favor of uint8_t. BUG=webrtc:5024 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1362503003 . Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/webrtc/base/helpers.cc
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07d09364b003e6738a02d9940aebab5d3814da6d |
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22-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Purge nss files and dependencies. This replaces https://codereview.webrtc.org/1313233005 which was reverted after triggering Chromium issues. The only difference is that we're cleaned up dependencies on use_openssl from the gyp file. Since https://codereview.chromium.org/1358913003 landed, this CL should cause no Chromium issues. BUG=webrtc:4497 Review URL: https://codereview.webrtc.org/1351503004 Cr-Commit-Position: refs/heads/master@{#10019}
/external/webrtc/webrtc/base/helpers.cc
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9eb1365939683cc5462a5359344148efb7d84f97 |
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05-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ ) Reason for revert: It looks like this broke the FYI bots. I tried updating libjingle_nacl.gyp, but the IOS build still failed because in Chrome it's configured to use NSS. See https://codereview.chromium.org/1316863012/. Original issue's description: > purge nss files and dependencies > > BUG=webrtc:4497 > > Committed: https://crrev.com/5647a2cf3db888195c928a1259d98f72f6ecbc15 > Cr-Commit-Position: refs/heads/master@{#9862} TBR=tommi@webrtc.org,kjellander@webrtc.org,torbjorng@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4497 Review URL: https://codereview.webrtc.org/1311843006 Cr-Commit-Position: refs/heads/master@{#9867}
/external/webrtc/webrtc/base/helpers.cc
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5647a2cf3db888195c928a1259d98f72f6ecbc15 |
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04-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
purge nss files and dependencies BUG=webrtc:4497 Review URL: https://codereview.webrtc.org/1313233005 Cr-Commit-Position: refs/heads/master@{#9862}
/external/webrtc/webrtc/base/helpers.cc
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979e0b30f1a5a8d2d7b7f1f5e9c6bc556b5b909b |
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16-Jun-2015 |
kjellander <kjellander@webrtc.org> |
Define uint64 and int64 using long long. This is to avoid typedef collisions with some compile configurations. For more info, see https://blogs.oracle.com/nike/entry/ilp64_lp64_llp64 http://www.unix.org/whitepapers/64bit.html BUG=4497 Review URL: https://codereview.webrtc.org/1186093004 Cr-Commit-Position: refs/heads/master@{#9451}
/external/webrtc/webrtc/base/helpers.cc
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469c2c04aae3e8446ba35f482adabd42800b41e1 |
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23-May-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Make Config::default_value leak instead of having an exit-time destructor. I wanted to use Config::Get in Chromium code, but it triggered the following warning: ../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors] static const T def; ^ ../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here return default_value<T>(); ^ I assume we don't hit this in webrtc because the warning is disabled. This also switches to the RTC_ prefix from the deprecated LIBJINGLE_. Needed due to this Chromium CL: https://codereview.chromium.org/1148843004/ R=andresp@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/53459004 Cr-Commit-Position: refs/heads/master@{#9268}
/external/webrtc/webrtc/base/helpers.cc
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67186fe00cc68cbe03aa66d17fb4962458ca96d2 |
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09-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/base Mostly this consists of marking functions with override when applicable, and moving function bodies from .h to .cc files. Not inlining virtual functions with simple bodies such as { return false; } strikes me as probably losing more in readability than we gain in binary size and compilation time, but I guess it's just like any other case where enabling a generally good warning forces us to write slightly worse code in a couple of places. BUG=163 R=kjellander@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47429004 Cr-Commit-Position: refs/heads/master@{#8656} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/helpers.cc
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d89b69aadeb9db67b7cc2de3300109d866c2a937 |
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06-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Fix WebRTC Win64 + BoringSSL build. There were many size_t to int conversions. RAND_poll and RAND_seed no longer do anything in BoringSSL, so fix that one by removing it. Use a checked_cast for the remaining ones. BUG=chromium:429039 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/helpers.cc
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f048872e915a3ee229044ec4bc541f6cbf9e4de1 |
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13-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/helpers.cc
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e9a604accd54ab14dbf98f99ccdcf3ae1c54d27c |
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13-May-2014 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457 > Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. > > BUG=N/A > R=andrew@webrtc.org, wu@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/12199004 TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/helpers.cc
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2c7d1b39b9374d2bc9bda4755fd4813db66a135c |
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12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. BUG=N/A R=andrew@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/helpers.cc
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