admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/pc/test
History
Niels Moller ec4a060a55 Revert "Delete leftover includes and declarations for MediaConstraintsInterface" 
This reverts commit a1e4ae23715867eca58488be307759ffa5901463.

Reason for revert: Breakage in downstream code still using constraints.

Original change's description:
> Delete leftover includes and declarations for MediaConstraintsInterface
> 
> Bug: webrtc:9239
> Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
> Reviewed-on: https://webrtc-review.googlesource.com/95721
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24442}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: Idbef4c57a0d3b82e94a431c5407a86c9fcd4be41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9239
Reviewed-on: https://webrtc-review.googlesource.com/96160
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24444}
2018-08-27 11:26:42 +00:00
..
androidtestinitializer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
androidtestinitializer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fakeaudiocapturemodule_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeaudiocapturemodule.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakeaudiocapturemodule.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakedatachannelprovider.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakepeerconnectionbase.h
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
fakepeerconnectionforstats.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakeperiodicvideosource.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
fakeperiodicvideotracksource.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
fakertccertificategenerator.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
fakesctptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
fakevideotrackrenderer.h
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
fakevideotracksource.h
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
framegeneratorcapturervideotracksource.h
Reland "Remove rtc::Optional alias and api:optional target"
2018-07-11 19:02:51 +00:00
mock_datachannel.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mock_peerconnection.h
Revert "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-08-27 11:26:42 +00:00
mock_rtpreceiverinternal.h
RtpReceiverInterface::stream_ids() added.
2018-07-11 10:14:56 +00:00
mock_rtpsenderinternal.h
Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface"
2018-05-15 15:51:02 +00:00
mockpeerconnectionobservers.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnectiontestwrapper.cc
Revert "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-08-27 11:26:42 +00:00
peerconnectiontestwrapper.h
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
rtcstatsobtainer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
testsdpstrings.h
By default, don't use SRTP_AES128_CM_SHA1_32 protection profile.
2018-04-05 23:43:07 +00:00