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.h
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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.h
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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.h
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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.h
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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.h
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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.h
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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.h
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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.h
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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.h
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