Steve Anton 111fdfd732 Refactor RtpSender to take the sender ID as a constructor argument ...

This change also standardizes the RtpSender to a single constructor
and moves the |track| and |stream_ids| arguments to setter methods.

Bug: webrtc:8734
Change-Id: I227a84868a80797f6cc2a1af6eec6d76da8ea159
Reviewed-on: https://webrtc-review.googlesource.com/84248
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23730}

2018-06-25 21:01:02 +00:00

..

BUILD.gn

Replace rtc::Optional with absl::optional in media, ortc, p2p

2018-06-16 07:09:59 +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

Fix build errors when rtc_use_builtin_sw_codecs is set to false.

2018-05-14 13:24:29 +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

Reland "Update RTCStatsCollector to work with RtpTransceivers"

2018-02-17 00:01:39 +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

Add pthatcher@ as OWNER of ortc/

2017-11-10 18:18:55 +00:00

rtpparametersconversion_unittest.cc

Replace rtc::Optional with absl::optional in media, ortc, p2p

2018-06-16 07:09:59 +00:00

rtpparametersconversion.cc

Replace rtc::Optional with absl::optional in media, ortc, p2p

2018-06-16 07:09:59 +00:00

rtpparametersconversion.h

Replace rtc::Optional with absl::optional in media, ortc, p2p

2018-06-16 07:09:59 +00:00

rtptransport_unittest.cc

OrtcFactoryInterface::Create(): Require caller to supply audio codec factories

2017-11-11 14:45:17 +00:00

rtptransportadapter.cc

Replace BundleFilter with RtpDemuxer in RtpTransport.

2018-04-14 00:57:11 +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

Reformat the WebRTC code base

2018-06-19 14:00:39 +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