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Anton Sukhanov 7940da0f2e Integration of media_transport in JSepTransportController 
Basic integration of media_transport in JSepTransportController.

- Creates media_transport if media transport factory provided in jsep transport controller configuration.
- Unittest that makes sure media_transport is created with correct caller or callee setting.
- Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests.

NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller.

NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile)

Bug: webrtc:9719
Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c
Reviewed-on: https://webrtc-review.googlesource.com/c/104400
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25096}
2018-10-10 18:25:25 +00:00
..
test
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
audiotrack.cc
Remove "using rtc::scoped_ptr" from audiotrack.cc.
2017-11-28 21:29:18 +00:00
audiotrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
BUILD.gn
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
builtin_video_bitrate_allocator_factory.cc
The default logic for creating video bitrate allocator.
2018-08-23 20:50:32 +00:00
builtin_video_bitrate_allocator_factory.h
The default logic for creating video bitrate allocator.
2018-08-23 20:50:32 +00:00
bundlefilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.h
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
2018-03-29 02:45:17 +00:00
channel_unittest.cc
Cleanup networkroute.h
2018-07-23 19:35:28 +00:00
channel.cc
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
channel.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
channelmanager_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
channelmanager.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
channelmanager.h
Use new TransportController implementation in PeerConnection.
2018-03-30 18:41:19 +00:00
createpeerconnectionfactory.cc
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
datachannel_unittest.cc
Removing warning suppression flags from pc/.
2018-07-04 10:35:27 +00:00
datachannel.cc
Use AsyncInvoker in DataChannel instead of MessageHandler
2018-08-31 22:48:20 +00:00
datachannel.h
Use AsyncInvoker in DataChannel instead of MessageHandler
2018-08-31 22:48:20 +00:00
DEPS
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
dtlssrtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
dtlssrtptransport.cc
Add a flag to actively reset the SRTP parameters
2018-06-12 20:32:00 +00:00
dtlssrtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
dtmfsender_unittest.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.h
Use AsyncInvoker in DtmfSender instead of MessageHandler
2018-08-31 22:49:26 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
[CFI] Fix cfi-icall failures
2017-12-12 09:27:07 +00:00
iceserverparsing_unittest.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepicecandidate.cc
Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h
2018-08-15 15:19:07 +00:00
jsepsessiondescription_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jsepsessiondescription.cc
Add CreateSessionDescription overload which takes a cricket::SessionDescription
2018-07-25 18:03:05 +00:00
jseptransport_unittest.cc
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
jseptransport.cc
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
jseptransport.h
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
jseptransportcontroller_unittest.cc
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
jseptransportcontroller.cc
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
jseptransportcontroller.h
Integration of media_transport in JSepTransportController
2018-10-10 18:25:25 +00:00
localaudiosource_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
Delete deprecated CreateAudioSource method, with constraints.
2018-06-04 08:19:30 +00:00
mediaconstraintsinterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
mediasession_unittest.cc
Make fewer copies when using StringBuilder.
2018-09-24 09:39:19 +00:00
mediasession.cc
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
mediasession.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamobserver.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
mediastreamtrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Adding shampson (me) as an owner to pc/ & api/.
2018-07-03 20:39:17 +00:00
peerconnection_bundle_unittest.cc
Add CreateIceCandidate overload which takes a cricket::Candidate
2018-07-23 23:49:44 +00:00
peerconnection_crypto_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnection_datachannel_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection_histogram_unittest.cc
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
peerconnection_ice_unittest.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
peerconnection_integrationtest.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
peerconnection_jsep_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection_media_unittest.cc
Add support for enabling simulcast in "Plan B" using MediaConstraints.
2018-08-24 09:55:59 +00:00
peerconnection_rampup_tests.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnection_rtp_unittest.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
peerconnection_signaling_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection.cc
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnection.h
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnectionendtoend_unittest.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
peerconnectionfactory_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectionfactory.cc
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnectionfactory.h
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnectioninterface_unittest.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
peerconnectionwrapper.h
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
remoteaudiosource.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtcpmuxfilter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
rtcstats_integrationtest.cc
getStats: add relayProtocol
2018-10-02 08:43:06 +00:00
rtcstatscollector_unittest.cc
getStats: add relayProtocol
2018-10-02 08:43:06 +00:00
rtcstatscollector.cc
getStats: add relayProtocol
2018-10-02 08:43:06 +00:00
rtcstatscollector.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtcstatstraversal_unittest.cc
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtcstatstraversal.cc
RTCStatsCollector::GetStatsReport() with optional selector argument.
2018-03-19 15:32:16 +00:00
rtcstatstraversal.h
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtpmediautils_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtpparametersconversion_unittest.cc
Parse two-byte header extensions.
2018-09-28 08:32:17 +00:00
rtpparametersconversion.cc
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpparametersconversion.h
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpreceiver.cc
Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed.
2018-10-10 16:45:39 +00:00
rtpreceiver.h
Add GetSources to VideoRtpReceiver
2018-09-27 10:00:40 +00:00
rtpsender.cc
Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed.
2018-10-10 16:45:39 +00:00
rtpsender.h
Add support for send_encodings parameters in addTransceiver
2018-10-01 22:56:30 +00:00
rtpsenderreceiver_unittest.cc
Enable End-to-End Encrypted Audio Payloads.
2018-10-04 22:08:34 +00:00
rtptransceiver.cc
[Unified Plan] Implement FiredDirection for RtpTransceiver
2018-07-17 23:56:04 +00:00
rtptransceiver.h
[Unified Plan] Implement FiredDirection for RtpTransceiver
2018-07-17 23:56:04 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
rtptransportinternal.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtptransportinternaladapter.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
rtptransporttestutil.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sdputils.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
sdputils.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
sessiondescription.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
sessiondescription.h
Add MSID signaling compatibility for Unified Plan endpoints
2018-02-01 21:17:41 +00:00
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Move base64.h to the proper location.
2018-07-23 15:40:36 +00:00
srtpfilter.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpsession_unittest.cc
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
srtpsession.cc
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtpsession.h
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtptestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
srtptransport.cc
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtptransport.h
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
statscollector_unittest.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
statscollector.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
transportstats.h
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
videocapturertracksource_unittest.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videotrack_unittest.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
videotrack.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
videotrack.h
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
videotracksource.cc
Delete deprecated VideoTrackSource constructor.
2018-06-11 19:15:10 +00:00
videotracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsdp_unittest.cc
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
webrtcsdp.cc
Make HasAttribute handle partial matching of attribute names.
2018-09-10 08:43:52 +00:00
webrtcsdp.h
Add MSID signaling compatibility for Unified Plan endpoints
2018-02-01 21:17:41 +00:00
webrtcsessiondescriptionfactory.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00