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 = |