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Moved retransmission rate limiter to Call class.

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Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.

Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
This commit is contained in:
Sebastian Jansson
2018-03-01 15:56:57 +01:00
committed by Commit Bot
parent 546d7f98a5
commit 45087cd23f
10 changed files with 12 additions and 26 deletions
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1
call/rtp_transport_controller_send_interface.h
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@ -99,7 +99,6 @@ class RtpTransportControllerSendInterface {
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
virtual int64_t GetPacerQueuingDelayMs() const = 0;
virtual int64_t GetFirstPacketTimeMs() const = 0;
virtual RateLimiter* GetRetransmissionRateLimiter() = 0;
virtual void EnablePeriodicAlrProbing(bool enable) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;