webrtc/call/test
Erik Språng 425d6aaa4c Add RtpPacketPacer interface for pacer control
The PacedSender is being reworked and will need an interface so we can
inject different implementations of it.

This CL introduces a new RtpPacketPacer interface inside the pacing
module. This interface handles the details of _how_ packets should be
paced, such as pacing rates/account for audio/max queue length etc.

The RtpPacketSender interface exposed from the rtp_rtcp module handles
only the actual sending of packets.

Some minor cleanups are included here.

Bug: webrtc:10809
Change-Id: I150b1a6262306d99e3f9d5f0b4afdb16a50e5ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28699}
2019-07-29 15:37:39 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
mock_rtp_transport_controller_send.h Add RtpPacketPacer interface for pacer control 2019-07-29 15:37:39 +00:00