From 15389c034d45eeaa81b4390a4959ddbc18e47540 Mon Sep 17 00:00:00 2001
From: nisse <nisse@webrtc.org>
Date: Tue, 24 Jan 2017 02:36:58 -0800
Subject: [PATCH] Drop pacer and retransmission_rate_limiter from
 RtpStreamReceiver constructor.

They were passed on via RtpRtcp::Configuration, but unused for a
receive only RtpRtcp module.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2639423007
Cr-Commit-Position: refs/heads/master@{#16234}
---
 webrtc/video/rtp_stream_receiver.cc  | 12 ++----------
 webrtc/video/rtp_stream_receiver.h   |  2 --
 webrtc/video/video_receive_stream.cc |  2 --
 3 files changed, 2 insertions(+), 14 deletions(-)

diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc
index d236085872..33fb50cc3a 100644
--- a/webrtc/video/rtp_stream_receiver.cc
+++ b/webrtc/video/rtp_stream_receiver.cc
@@ -51,9 +51,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
     RemoteBitrateEstimator* remote_bitrate_estimator,
-    RtpPacketSender* paced_sender,
-    TransportSequenceNumberAllocator* transport_sequence_number_allocator,
-    RateLimiter* retransmission_rate_limiter) {
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
   RtpRtcp::Configuration configuration;
   configuration.audio = false;
   configuration.receiver_only = true;
@@ -63,7 +61,6 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
-  configuration.paced_sender = paced_sender;
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = nullptr;
@@ -72,7 +69,6 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
   configuration.send_packet_observer = nullptr;
   configuration.bandwidth_callback = nullptr;
   configuration.transport_feedback_callback = nullptr;
-  configuration.retransmission_rate_limiter = retransmission_rate_limiter;
 
   std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
   rtp_rtcp->SetSendingStatus(false);
@@ -89,13 +85,11 @@ RtpStreamReceiver::RtpStreamReceiver(
     RemoteBitrateEstimator* remote_bitrate_estimator,
     Transport* transport,
     RtcpRttStats* rtt_stats,
-    PacedSender* paced_sender,
     PacketRouter* packet_router,
     VieRemb* remb,
     const VideoReceiveStream::Config* config,
     ReceiveStatisticsProxy* receive_stats_proxy,
     ProcessThread* process_thread,
-    RateLimiter* retransmission_rate_limiter,
     NackSender* nack_sender,
     KeyFrameRequestSender* keyframe_request_sender,
     video_coding::OnCompleteFrameCallback* complete_frame_callback,
@@ -123,9 +117,7 @@ RtpStreamReceiver::RtpStreamReceiver(
                                     rtt_stats,
                                     receive_stats_proxy,
                                     remote_bitrate_estimator_,
-                                    paced_sender,
-                                    packet_router,
-                                    retransmission_rate_limiter)),
+                                    packet_router)),
       complete_frame_callback_(complete_frame_callback),
       keyframe_request_sender_(keyframe_request_sender),
       timing_(timing) {
diff --git a/webrtc/video/rtp_stream_receiver.h b/webrtc/video/rtp_stream_receiver.h
index b1e1db4b14..07ad4b7e01 100644
--- a/webrtc/video/rtp_stream_receiver.h
+++ b/webrtc/video/rtp_stream_receiver.h
@@ -69,13 +69,11 @@ class RtpStreamReceiver : public RtpData,
       RemoteBitrateEstimator* remote_bitrate_estimator,
       Transport* transport,
       RtcpRttStats* rtt_stats,
-      PacedSender* paced_sender,
       PacketRouter* packet_router,
       VieRemb* remb,
       const VideoReceiveStream::Config* config,
       ReceiveStatisticsProxy* receive_stats_proxy,
       ProcessThread* process_thread,
-      RateLimiter* retransmission_rate_limiter,
       NackSender* nack_sender,
       KeyFrameRequestSender* keyframe_request_sender,
       video_coding::OnCompleteFrameCallback* complete_frame_callback,
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 183f72b537..174d352564 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -212,13 +212,11 @@ VideoReceiveStream::VideoReceiveStream(
               UseSendSideBwe(config_)),
           &transport_adapter_,
           call_stats_->rtcp_rtt_stats(),
-          congestion_controller_->pacer(),
           packet_router,
           remb,
           &config_,
           &stats_proxy_,
           process_thread_,
-          congestion_controller_->GetRetransmissionRateLimiter(),
           this,  // NackSender
           this,  // KeyFrameRequestSender
           this,  // OnCompleteFrameCallback