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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
c6ce9c59385102e766913dcfc5eb69cb5cf135ba
/
api
/
mediastreaminterface.h
c6ce9c5
New file api/video/BUILD.gn
by Niels Möller
· 6 years ago
13b8bad
Final name changing of MediaStreamInterface.label() to id().
by Seth Hampson
· 6 years ago
845e878
Name change from stream label to stream id for spec compliance.
by Seth Hampson
· 7 years ago
2a5ce2b
Fix clang style errors in rtp_rtcp and dependant targets
by Danil Chapovalov
· 7 years ago
9e19403
Move videosourceinterface to api.
by Patrik Höglund
· 7 years ago
be214a2
Move videosinkinterface.h to common_video to solve a circular dep.
by Patrik Höglund
· 7 years ago
21eb9fc
Make the old GetStats interface on AudioProcessorInterface impure.
by Ivo Creusen
· 7 years ago
3a23374
Reland "Remove the aec_quality_min metric."
by Gustaf Ullberg
· 7 years ago
a3fad93
Revert "Remove the aec_quality_min metric."
by Mirko Bonadei
· 7 years ago
99b1bd1
Remove the aec_quality_min metric.
by Gustaf Ullberg
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
ae02609
Add parallel stats interface with optional stats to APM.
by Ivo Creusen
· 7 years ago
e2d6a06
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
1af3d82
Revert "Reland "Clean up libjingle API dependencies.""
by Henrik Kjellander
· 7 years ago
9185aca
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
581df61
Revert "Reland "Clean up libjingle API dependencies.""
by Patrik Höglund
· 7 years ago
5117b04
Reland "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
7bcfc3b
Revert "Clean up libjingle API dependencies."
by Patrik Höglund
· 7 years ago
57fb315
Clean up libjingle API dependencies.
by Patrik Höglund
· 7 years ago
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
· 7 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
· 7 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
· 7 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
· 7 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
· 7 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
· 7 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
· 7 years ago
713a3bb
Delete deprecated and transitional stuff related to video frame refactoring.
by nisse
· 7 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
· 8 years ago
2a8a78c
Add AEC filter divergence metric to StatsCollector.
by Minyue
· 8 years ago
efc3858
Remove deprecated MediaStreamTrackInterface::set_state
by perkj
· 8 years ago
fcc640f
Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector,
by nisse
· 8 years ago
c0d31e9
Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool>
by Per
· 8 years ago
7ca142e
ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy.
by perkj
· 8 years ago
d61bf80
Removed MediaStreamTrackInterface::set_state
by perkj
· 8 years ago
8f59762
Delete VideoRendererInterface.
by Niels Möller
· 8 years ago
c8f952d
Propagate MediaStreamSource state to video tracks the same way as audio.
by perkj
· 8 years ago
f0dcfe2
Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource.
by perkj
· 8 years ago
0d3eef2
Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it.
by perkj
· 8 years ago
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 8 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
· 9 years ago
f9a75d9
Revert "Add concept of whether video renderer supports rotation."
by guoweis@webrtc.org
· 9 years ago
0ad4893
Add concept of whether video renderer supports rotation.
by guoweis@webrtc.org
· 9 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
· 10 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