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Henrik Boström 199e27bb63 RtpReceiverInterface::stream_ids() added. 
This is the first step to removing streams from third_party/webrtc.
RtpReceiverInterface::streams() will have to be removed separately.
See https://crbug.com/webrtc/9480 for more information.

Bug: webrtc:9480
Change-Id: I6f9e6ddcda5e2245cc601d2cc6205b7b363f73ef
Reviewed-on: https://webrtc-review.googlesource.com/86840
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23929}
2018-07-11 10:14:56 +00:00
..
BUILD.gn
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
DEPS
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ortcfactory_integrationtest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
ortcfactory_unittest.cc
OrtcFactoryInterface::Create(): Require caller to supply audio codec factories
2017-11-11 14:45:17 +00:00
ortcfactory.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
ortcfactory.h
Delete ortc methods using cricket::VideoCapturer.
2018-03-22 08:55:24 +00:00
ortcrtpreceiver_unittest.cc
OrtcFactoryInterface::Create(): Require caller to supply audio codec factories
2017-11-11 14:45:17 +00:00
ortcrtpreceiveradapter.cc
RtpReceiverInterface::stream_ids() added.
2018-07-11 10:14:56 +00:00
ortcrtpreceiveradapter.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ortcrtpsender_unittest.cc
OrtcFactoryInterface::Create(): Require caller to supply audio codec factories
2017-11-11 14:45:17 +00:00
ortcrtpsenderadapter.cc
Refactor RtpSender to take the sender ID as a constructor argument
2018-06-25 21:01:02 +00:00
ortcrtpsenderadapter.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Removing deadbeef@ from OWNERS files.
2018-07-02 00:40:38 +00:00
rtptransport_unittest.cc
OrtcFactoryInterface::Create(): Require caller to supply audio codec factories
2017-11-11 14:45:17 +00:00
rtptransportadapter.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtptransportadapter.h
Replace BundleFilter with RtpDemuxer in RtpTransport.
2018-04-14 00:57:11 +00:00
rtptransportcontroller_unittest.cc
Use new TransportController implementation in PeerConnection.
2018-03-30 18:41:19 +00:00
rtptransportcontrolleradapter.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtptransportcontrolleradapter.h
Use new TransportController implementation in PeerConnection.
2018-03-30 18:41:19 +00:00
srtptransport_unittest.cc
Use new TransportController implementation in PeerConnection.
2018-03-30 18:41:19 +00:00
testrtpparameters.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
testrtpparameters.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00