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webrtc_m130/media/base
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Ilya Nikolaevskiy ca536d4692 Revert "Enable simulcast screenshare by default" 
This reverts commit d43c692ba7f53b5576a494c0343bc7a4bb36831b.

Reason for revert: Breaks chromium unit tests

Original change's description:
> Enable simulcast screenshare by default
> 
> Bug: chromium:690537
> Change-Id: I8b713a9c4d9d5d1a5cf13dff607cc25806aceed2
> Reviewed-on: https://webrtc-review.googlesource.com/87560
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24003}

TBR=ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: I55b952519458bb9ab49cf6377601d7420e71d086
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:690537
Reviewed-on: https://webrtc-review.googlesource.com/89080
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24005}
2018-07-17 12:47:55 +00:00
..
adaptedvideotracksource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
adaptedvideotracksource.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
audiosource.h
…
codec_unittest.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.cc
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00
codec.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
device.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeframesource.cc
Refactor PeerConnectionIntegrationTest to not use cricket::VideoCapturer
2018-05-11 09:22:20 +00:00
fakeframesource.h
Refactor PeerConnectionIntegrationTest to not use cricket::VideoCapturer
2018-05-11 09:22:20 +00:00
fakemediaengine.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakenetworkinterface.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertp.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakevideocapturer.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakevideocapturer.h
Introduce TaskQueueForTest.
2018-02-14 15:32:49 +00:00
fakevideorenderer.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
h264_profile_level_id.cc
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
h264_profile_level_id.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
mediachannel.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
mediachannel.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconfig.h
Remove wire-up for cancelled experement on VAAPI VP8 encoding
2018-06-15 10:04:07 +00:00
mediaconstants.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediaconstants.h
Add RRTR parameter to media engine and pass it to video receive stream
2018-04-06 11:15:12 +00:00
mediaengine.cc
Add Rtcp parameters for PeerConnection senders
2018-05-28 09:28:59 +00:00
mediaengine.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtpdataengine.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtputils.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamparams.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
testutils.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turnutils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
turnutils.cc
…
turnutils.h
…
videoadapter_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
videoadapter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videoadapter.h
Replace rtc::Optional with absl::optional in media, ortc, p2p
2018-06-16 07:09:59 +00:00
videobroadcaster_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videobroadcaster.h
New file api/video/BUILD.gn
2018-05-14 06:57:38 +00:00
videocapturer_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocapturer.cc
Revert "Enable simulcast screenshare by default"
2018-07-17 12:47:55 +00:00
videocapturer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocapturerfactory.h
…
videocommon_unittest.cc
…
videocommon.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videocommon.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
videosourcebase.cc
Enable and fix chromium clang warnings in sdk/android targets.
2018-04-05 11:22:03 +00:00
videosourcebase.h
Move VideoStreamEncoderInterface to api/.
2018-05-21 19:50:37 +00:00
vp9_profile.cc
Reland "Remove rtc::Optional alias and api:optional target"
2018-07-11 19:02:51 +00:00
vp9_profile.h
Add VP9 profile negotiation to SDP
2018-07-02 23:38:41 +00:00