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} |
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| .. | ||
| mock_audio_send_stream.h | ||
| mock_bitrate_allocator.h | ||
| mock_rtp_packet_sink_interface.h | ||
| mock_rtp_transport_controller_send.h | ||