| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
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/talk/media/base/fakevideocapturer.h
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| b09b660c53ff2c499d149e05e5c435f5057273fc |
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01-Oct-2015 |
magjed <magjed@webrtc.org> |
Remove cricket::VideoFrame::Set/GetElapsedTime()
This CL is a baby step towards consolidating the timestamps in cricket::VideoFrame and webrtc::VideoFrame, so that we can unify the frame classes in the future.
The elapsed time functionality is not really used. If a video sink wants to know the elapsed time since the first frame they can store the first timestamp themselves and calculate the time delta to later frames. This is already done in all video sinks that need the elapsed time. Having redundant timestamps in the frame classes is confusing and error prone.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1324263004
Cr-Commit-Position: refs/heads/master@{#10131}
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 1cb121dea478a4bb4f88e76cf92719e2853543cf |
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14-Sep-2015 |
pbos <pbos@webrtc.org> |
Reset frame timestamp epoch for new capturers.
Incoming frames usually have an epoch of time since the capturer was
created or similar, not any fixed-time epoch. As such, setting a new
capturer resulted in delivering frames with older timestamps which
caused these frames to be dropped before encoding.
BUG=webrtc:4994
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345473002
Cr-Commit-Position: refs/heads/master@{#9934}
/external/webrtc/talk/media/base/fakevideocapturer.h
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| c27d89fdc6b33846ff06e8ca8bd03119d05c6530 |
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16-Jul-2015 |
qiangchen <qiangchen@chromium.org> |
Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 1226e926e6104322d9b99026b98f515cb4d40fd4 |
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11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8337
Committed: https://code.google.com/p/webrtc/source/detail?r=8338
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8339}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8339 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| dc7b02277cc1666dfc13b636c2ecfe53b12c9d2a |
|
11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8337
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8338}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 20e8f227664a6747cea11e1fc1de4c018ebcc8e9 |
|
11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8337}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8337 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 35c1ace18532b50ff274f65b1369889baefca319 |
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13-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats
> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
>
> BUG=3936
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/30039004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 52da44b7e61bb6cf3f36c8c4f8de1a888e2814bb |
|
13-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
BUG=3936
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 4f0d401faecf5d8a4c82e6e2223651ef13ad8e31 |
|
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72682155-> 72785180
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
|
| 8e885990aed58a26ae1b27dd7547536393879a7c |
|
05-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72566057-> 72591796
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 3bc48247b7d009f708550cd5c0470038f9045d08 |
|
01-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72403605-> 72407428
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 97077a3ab27259164eb121034b6e0ebe9ba592df |
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25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
|
| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
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22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1848004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
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| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/base/fakevideocapturer.h
|