Erik Språng aa59eca891 Move RtpPacketSender and merge it with RtpPacketPacer. ... This interface is intended to only handle packet-sending parts of the paced sender. See https://webrtc-review.googlesource.com/c/src/+/145212 for context Bug: webrtc:10809 Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28662}		2019-07-24 13:28:21 +00:00
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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	Format almost everything.	2019-07-08 13:45:15 +00:00
mock_rtp_transport_controller_send.h	Move RtpPacketSender and merge it with RtpPacketPacer.	2019-07-24 13:28:21 +00:00