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webrtc/call/test
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Sebastian Jansson 2997ec9a7a Removes unused keep-alive from RtpTransportControllerSend. 
This prepares for future cleanup of how RtpTransportControllerSend is
used.

Bug: webrtc:10365
Change-Id: Idefc7e60f83819627c83b397949c8434d93491b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26923}
2019-03-01 12:15:54 +00:00
..
mock_audio_send_stream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_bitrate_allocator.h Declares BitrateAllocator methods const. 2018-10-24 15:26:36 +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 Removes unused keep-alive from RtpTransportControllerSend. 2019-03-01 12:15:54 +00:00