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webrtc_m130/pc/test
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Niklas Enbom 82c71af262 Revert "Modernize rtc::SSLCertificate" 
This reverts commit 55cd3ac804811e02b9b14026c683f9b30ea0c0bb.

Reason for revert: Breaks Chrome compile: https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8932588150164377824/+/steps/compile__with_patch_/0/stdout 

Original change's description:
> Modernize rtc::SSLCertificate
> 
> Bug: webrtc:9860
> Change-Id: Idfce546ded500d957397c5bd873200565d3e6b64
> Reviewed-on: https://webrtc-review.googlesource.com/c/105280
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25150}

TBR=steveanton@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9860
Change-Id: I4ff090f2612252cd656a34a0181aff81488c6edf
Reviewed-on: https://webrtc-review.googlesource.com/c/105946
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25182}
2018-10-15 17:31:05 +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
Revert "Modernize rtc::SSLCertificate"
2018-10-15 17:31:05 +00:00
fakeperiodicvideosource.h
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +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_rtpreceiverinternal.h
Injects FrameEncryptorInterface into RtpSender.
2018-08-30 00:33:54 +00:00
mock_rtpsenderinternal.h
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
mockpeerconnectionobservers.h
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnectiontestwrapper.cc
Add sanity checks to UpdateDelayStatistics and patch unit tests.
2018-09-24 23:13:02 +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