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Unified Diff: webrtc/video/video_receive_stream.cc

Issue 2181383002: Add NACK rate throttling for audio channels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 5 months ago
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Index: webrtc/video/video_receive_stream.cc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 609a789fa2b530a2d65ecdb29c10fe98ee1b66eb..bc51269b85916e1daf085f17b96a6cfb4bb59734 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -14,6 +14,7 @@
#include <set>
#include <string>
+#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
@@ -162,17 +163,19 @@ VideoReceiveStream::VideoReceiveStream(
call_stats_(call_stats),
video_receiver_(clock_, nullptr, this, this, this),
stats_proxy_(&config_, clock_),
- rtp_stream_receiver_(&video_receiver_,
- congestion_controller_->GetRemoteBitrateEstimator(
- UseSendSideBwe(config_)),
- &transport_adapter_,
- call_stats_->rtcp_rtt_stats(),
- congestion_controller_->pacer(),
- congestion_controller_->packet_router(),
- remb,
- &config_,
- &stats_proxy_,
- process_thread_),
+ rtp_stream_receiver_(
+ &video_receiver_,
+ congestion_controller_->GetRemoteBitrateEstimator(
+ UseSendSideBwe(config_)),
+ &transport_adapter_,
+ call_stats_->rtcp_rtt_stats(),
+ congestion_controller_->pacer(),
+ congestion_controller_->packet_router(),
+ remb,
+ &config_,
+ &stats_proxy_,
+ process_thread_,
+ congestion_controller_->GetRetransmissionRateLimiter()),
vie_sync_(&video_receiver_) {
LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
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