webrtc/api
Anton Sukhanov da65ed2adc Reland "Propagate media transport to media channel."
This reverts commit 37cf2455a4.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Revert "Propagate media transport to media channel."
> 
> This reverts commit 8c16f745ab.
> 
> Reason for revert: Breaks downstream project
> 
> Original change's description:
> > Propagate media transport to media channel.
> > 
> > 1. Pass media transport factory to JSEP transport controller.
> > 2. Pass media transport to voice media channel.
> > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> > 
> > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > Bug: webrtc:9719
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > Cr-Commit-Position: refs/heads/master@{#25152}
> 
> TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:9719
> Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25154}

TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com

Change-Id: I505ff3451eae81573531faef155ff35d7f894022
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/106500
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25220}
2018-10-16 18:22:44 +00:00
..
audio Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
audio_codecs Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
call Adds flags indicating presence in allocation and feedback per packet. 2018-10-09 18:24:38 +00:00
crypto Removes backwards compatability CryptoOptions support. 2018-10-12 18:22:23 +00:00
ortc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
stats Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
test Reland "Propagate media transport to media channel." 2018-10-16 18:22:44 +00:00
transport Implement (mostly) standards-compliant RTCIceTransportState. 2018-10-10 08:10:16 +00:00
units Using more specific dependencies in rtc_base. 2018-10-16 12:00:09 +00:00
video Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
video_codecs Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
array_view.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
array_view_unittest.cc ArrayView, adding ctor for fixed-size views of const(expr) std::array. 2018-05-15 13:49:02 +00:00
asyncresolverfactory.h Support domain name ICE candidates 2018-08-24 04:54:43 +00:00
audio_options.cc Remove AECM comfort noise setting from API 2018-10-16 09:42:16 +00:00
audio_options.h Remove AECM comfort noise setting from API 2018-10-16 09:42:16 +00:00
bitrate_constraints.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
BUILD.gn Implement test class LoopbackMediaTransport 2018-10-16 09:21:28 +00:00
candidate.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
candidate.h Export symbols needed by the Chromium component build (part 2). 2018-10-15 19:52:31 +00:00
cryptoparams.h Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" 2018-10-11 23:09:07 +00:00
datachannelinterface.cc Enabling clang::find_bad_constructs for libjingle_peerconnection_api. 2018-07-19 09:17:10 +00:00
datachannelinterface.h Enabling clang::find_bad_constructs for libjingle_peerconnection_api. 2018-07-19 09:17:10 +00:00
DEPS Moves GoogCC factory to API. 2018-10-10 06:11:36 +00:00
dtmfsenderinterface.h Add "tones remaining" argument to DTMF ontonechange callback 2018-09-07 17:29:37 +00:00
fec_controller.h Revert "Revert "Enables PeerConnectionFactory using external fec controller"" 2018-02-20 12:41:55 +00:00
jsep.cc Move SdpType from/to string definition close to declaration. 2018-10-12 09:59:40 +00:00
jsep.h Export symbols needed by the Chromium component build (part 2). 2018-10-15 19:52:31 +00:00
jsepicecandidate.cc Enable clang::find_bad_constructs for sdk/android (part 1/2). 2018-07-20 21:35:40 +00:00
jsepicecandidate.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
jsepsessiondescription.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
media_transport_interface.cc Pass MediaTransportFactory to PeerConnectionFactory. 2018-10-08 18:11:06 +00:00
media_transport_interface.h Pass MediaTransportFactory to PeerConnectionFactory. 2018-10-08 18:11:06 +00:00
mediaconstraintsinterface.cc Removing the intelligibility enhancer. 2018-08-30 21:29:57 +00:00
mediaconstraintsinterface.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
mediastreaminterface.cc Final name changing of MediaStreamInterface.label() to id(). 2018-03-14 20:30:52 +00:00
mediastreaminterface.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
mediastreamproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastreamtrackproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediatypes.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediatypes.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
notifier.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Adding shampson (me) as an owner to pc/ & api/. 2018-07-03 20:39:17 +00:00
peerconnectionfactoryproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectioninterface.cc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
peerconnectioninterface.h Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
peerconnectionproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
proxy.cc Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
proxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
refcountedbase.h New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
rtcerror.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror.h Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror_unittest.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtceventlogoutput.h Move RtcEventLogOutput to api/ 2017-10-06 13:58:14 +00:00
rtp_headers.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtp_headers.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtpparameters.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtpparameters.h Add support for sending RTP two-byte header extensions. 2018-10-05 08:45:52 +00:00
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpreceiverinterface.cc Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpreceiverinterface.h Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpsenderinterface.cc Add support for send_encodings parameters in addTransceiver 2018-10-01 22:56:30 +00:00
rtpsenderinterface.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtptransceiverinterface.cc Add dummy implementation for SetCodecPReferences. 2018-09-24 18:26:10 +00:00
rtptransceiverinterface.h Add dummy implementation for SetCodecPReferences. 2018-09-24 18:26:10 +00:00
setremotedescriptionobserverinterface.h Reland "SetRemoteDescriptionObserverInterface added." 2017-11-23 19:59:48 +00:00
statstypes.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
statstypes.h Revert "Add framesRendered to StatsReport" 2018-07-27 14:53:07 +00:00
turncustomizer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
umametrics.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
videosourceproxy.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00