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Unified Diff: webrtc/video/video_send_stream.cc

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 4 years, 5 months ago
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Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index eeb9580b6b707e1c454491fc3339c9508eb5215a..ce79f61568c4734fe459e4a3d9566cdce1c13607 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -52,6 +52,7 @@ std::vector<RtpRtcp*> CreateRtpRtcpModules(
SendStatisticsProxy* stats_proxy,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
+ RateLimiter* retransmission_rate_limiter,
size_t num_modules) {
RTC_DCHECK_GT(num_modules, 0u);
RtpRtcp::Configuration configuration;
@@ -73,6 +74,7 @@ std::vector<RtpRtcp*> CreateRtpRtcpModules(
configuration.send_side_delay_observer = stats_proxy;
configuration.send_packet_observer = send_delay_stats;
configuration.event_log = event_log;
+ configuration.retransmission_rate_limiter = retransmission_rate_limiter;
std::vector<RtpRtcp*> modules;
for (size_t i = 0; i < num_modules; ++i) {
@@ -426,6 +428,7 @@ VideoSendStream::VideoSendStream(
&stats_proxy_,
send_delay_stats,
event_log,
+ congestion_controller_->GetRetransmissionRateLimiter(),
config_.rtp.ssrcs.size())),
payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type),
input_(&encoder_wakeup_event_,
@@ -869,7 +872,6 @@ void VideoSendStream::SignalNetworkState(NetworkState state) {
uint32_t VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt) {
- payload_router_.SetTargetSendBitrate(bitrate_bps);
// Get the encoder target rate. It is the estimated network rate -
// protection overhead.
uint32_t encoder_target_rate_bps =

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