commit aa59eca89198420f73184ba7c0576b080c817684
author Erik Språng <sprang@webrtc.org> Wed Jul 24 14:52:55 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Jul 24 13:28:21 2019 +0000
tree 198aaa1b43b2174b4a932bbd2f49016af16bcb3d
parent 55c4a42099d95a751d9610c773e448f4a755f2a6

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}

call/rtp_transport_controller_send_interface.h

diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 0178758..39358d5 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -27,7 +27,7 @@
 #include "logging/rtc_event_log/rtc_event_log.h"
 #include "modules/rtp_rtcp/include/report_block_data.h"
 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 
@@ -119,7 +119,7 @@
   virtual NetworkStateEstimateObserver* network_state_estimate_observer() = 0;
   virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
 
-  virtual RtpPacketPacer* packet_sender() = 0;
+  virtual RtpPacketSender* packet_sender() = 0;
 
   // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
   // settings.
@@ -160,6 +160,8 @@
 
   virtual void OnTransportOverheadChanged(
       size_t transport_overhead_per_packet) = 0;
+
+  virtual void AccountForAudioPacketsInPacedSender(bool account_for_audio) = 0;
 };
 
 }  // namespace webrtc