Log - 5a6aa4f05df7ea853afe83e6264f2e992c6ebb2e - platform/external/webrtc

92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/mediastreaminterface.h]
8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 7 years ago
f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
f93752a Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 8 years ago
61b22dd Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 8 years ago
3870a07 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 8 years ago
6e6a485 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 8 years ago
d71ebd7 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 8 years ago
aec49d2 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 8 years ago
713a3bb Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
8d60a94 Replace NULL with nullptr or null in webrtc/api/. by deadbeef · 8 years ago
b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
af91689 Move VideoFrame and related declarations to webrtc/api/video. by nisse · 8 years ago
9baddf2 Replace basictypes.h with stddef.h for size_t. by pbos · 8 years ago
5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
7341ab8 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
859e861 Remove stop method from VideoTrackSourceInterface. by sakal · 8 years ago
a973f95 Remove restart method from VideoTrackSourceInterface. by sakal · 8 years ago
5d58333 Fix VideoFrame inclusion in mediastreaminterface.h by perkj · 8 years ago
2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
2a8a78c Add AEC filter divergence metric to StatsCollector. by Minyue · 9 years ago
efc3858 Remove deprecated MediaStreamTrackInterface::set_state by perkj · 9 years ago
fcc640f Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 9 years ago
c0d31e9 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 9 years ago
7ca142e ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy. by perkj · 9 years ago
d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 9 years ago
8f59762 Delete VideoRendererInterface. by Niels Möller · 9 years ago
c8f952d Propagate MediaStreamSource state to video tracks the same way as audio. by perkj · 9 years ago
f0dcfe2 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. by perkj · 9 years ago
0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 9 years ago
a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
db25d2e Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface. by nisse · 9 years ago
b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/mediastreaminterface.h]
8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 9 years ago
e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
6a062bd Deleted method AudioTrackInterface::GetRenderer. by nisse · 9 years ago
2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 9 years ago
6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
c2db810 Remove VideoRendererInterface::CanApplyRotation() by Magnus Jedvert · 9 years ago
dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
00c509a Add concept of whether video renderer supports rotation. by guoweis@webrtc.org · 10 years ago
f9a75d9 Revert "Add concept of whether video renderer supports rotation." by guoweis@webrtc.org · 10 years ago
0ad4893 Add concept of whether video renderer supports rotation. by guoweis@webrtc.org · 10 years ago
5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 11 years ago
40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 11 years ago
b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago
67ee6b9 Update talk to 60923971 by mallinath@webrtc.org · 11 years ago
967bfff Update talk to 52534915. by wu@webrtc.org · 11 years ago
32001ef PeerConnection shutdown-time fixes by fischman@webrtc.org · 11 years ago
1e09a71 Update talk folder to revision=49952949 by henrike@webrtc.org · 11 years ago
28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago