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Steve Anton 9158ef6575 Reland "Add AddTransceiver and GetTransceivers to PeerConnection" 
This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80.

Original change's description:
> Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
> 
> This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.
> 
> Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
> 
> Original change's description:
> > Add AddTransceiver and GetTransceivers to PeerConnection
> > 
> > WebRTC 1.0 has added the transceiver API to PeerConnection. This
> > is the first step towards exposing this to WebRTC consumers. For
> > now, transceivers can be added and fetched but there is not yet
> > support for creating offers/answers or setting local/remote
> > descriptions. That support ("Unified Plan") will be added in
> > follow-up CLs.
> > 
> > The transceiver API is currently only available if the application
> > opts in by specifying the kUnifiedPlan SDP semantics when creating
> > the PeerConnection.
> > 
> > Bug: webrtc:7600
> > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> > Reviewed-on: https://webrtc-review.googlesource.com/23880
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20896}
> 
> TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
> 
> Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7600
> Reviewed-on: https://webrtc-review.googlesource.com/26400
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20897}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org

Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 22:27:49 +00:00
..
audio
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
audio_codecs
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 11:21:47 +00:00
call
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ortc
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
stats
Add network_type to local RTCIceCandidateStats
2017-11-21 19:58:37 +00:00
test
Add StereoCodecAdapter classes
2017-10-31 06:39:52 +00:00
video
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 11:21:47 +00:00
video_codecs
Optional: Use nullopt and implicit construction in /api
2017-11-24 13:18:09 +00:00
array_view_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
array_view.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
BUILD.gn
Use the new AudioProcessing statistics everywhere.
2017-11-24 18:17:39 +00:00
candidate.cc
Fix clang style warnings in api/candidate.h
2017-10-26 23:22:18 +00:00
candidate.h
Fix clang style warnings in api/candidate.h
2017-10-26 23:22:18 +00:00
cryptoparams.h
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
datachannel.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
datachannelinterface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Use the new AudioProcessing statistics everywhere.
2017-11-24 18:17:39 +00:00
dtmfsenderinterface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fakemetricsobserver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
fakemetricsobserver.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsep.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepicecandidate.h
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
jsepsessiondescription.h
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
mediaconstraintsinterface.cc
Optional: Use nullopt and implicit construction in /api
2017-11-24 13:18:09 +00:00
mediaconstraintsinterface.h
Android: Generate JNI code for MediaConstraints
2017-11-24 11:03:19 +00:00
mediastream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreaminterface.cc
Use the new AudioProcessing statistics everywhere.
2017-11-24 18:17:39 +00:00
mediastreaminterface.h
Use the new AudioProcessing statistics everywhere.
2017-11-24 18:17:39 +00:00
mediastreamproxy.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamtrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamtrackproxy.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediatypes.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
notifier.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
optional_unittest.cc
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
optional.cc
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
optional.h
Added nullopt and implicit construction to rtc::Optional
2017-11-15 15:50:11 +00:00
OWNERS
Make hbos@webrtc.org OWNER of peerconnection*.
2017-11-13 12:27:29 +00:00
peerconnectionfactoryproxy.h
Remove dead version of StartRtcEventLog
2017-10-06 15:18:24 +00:00
peerconnectioninterface.h
Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
2017-11-27 22:27:49 +00:00
peerconnectionproxy.h
Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
2017-11-27 22:27:49 +00:00
proxy.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
proxy.h
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
refcountedbase.h
New classes RefCounter and RefCountedBase.
2017-10-23 11:46:47 +00:00
rtcerror_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcerror.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcerror.h
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtceventlogoutput.h
Move RtcEventLogOutput to api/
2017-10-06 13:58:14 +00:00
rtpparameters_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpparameters.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpparameters.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpreceiverinterface.h
Add streams() to RtpReceiverInterface and implementations.
2017-11-22 09:04:17 +00:00
rtpsender.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpsenderinterface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtptransceiverinterface.h
Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
2017-11-27 22:27:49 +00:00
setremotedescriptionobserverinterface.h
Reland "SetRemoteDescriptionObserverInterface added."
2017-11-23 19:59:48 +00:00
statstypes.cc
Add stats googHasEnteredLowResolution.
2017-11-17 13:02:07 +00:00
statstypes.h
Add stats googHasEnteredLowResolution.
2017-11-17 13:02:07 +00:00
streamcollection.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
turncustomizer.h
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
umametrics.cc
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
umametrics.h
Enable the clang style plugin in primary p2p/ target
2017-11-01 00:19:05 +00:00
videosourceproxy.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videotracksource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
webrtcsdp.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00