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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moved rate limiter and addressed comments Created 4 years, 6 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index be8ab34a2771117680367ff7e1e4653ec4c3ab01..05108af80d5a0d7da373818791f6315e184a4398 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -63,7 +63,8 @@ RtpRtcp::Configuration::Configuration()
send_frame_count_observer(nullptr),
send_side_delay_observer(nullptr),
event_log(nullptr),
- send_packet_observer(nullptr) {}
+ send_packet_observer(nullptr),
+ nack_rate_limiter(nullptr) {}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
if (configuration.clock) {
@@ -89,7 +90,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
configuration.send_frame_count_observer,
configuration.send_side_delay_observer,
configuration.event_log,
- configuration.send_packet_observer),
+ configuration.send_packet_observer,
+ configuration.nack_rate_limiter),
rtcp_sender_(configuration.audio,
configuration.clock,
configuration.receive_statistics,

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