Stefan Holmer 5ed25af448 Properly clean up RtpVideoSender. ...

Bug: webrtc:9517
Change-Id: I625c132907bd178f62c8b99f4b2407c75aa7e947
Reviewed-on: https://webrtc-review.googlesource.com/89382
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24022}

2018-07-18 14:15:07 +00:00

..

fake_network_pipe_unittest.cc

Moving demux from FakeNetworkPipe to DirectTransport.

2018-04-25 10:13:03 +00:00

mock_audio_send_stream.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

mock_bitrate_allocator.h

Adds mock bitrate allocator.

2018-04-19 14:41:42 +00:00

mock_rtp_packet_sink_interface.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

mock_rtp_transport_controller_send.h

Properly clean up RtpVideoSender.

2018-07-18 14:15:07 +00:00