| Index: webrtc/video/video_send_stream.cc
|
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
|
| index 3e95cb02956fc8d350703ca617704322fe513ba0..a85cf314a24c3579d7176e388c15464f1891d02c 100644
|
| --- a/webrtc/video/video_send_stream.cc
|
| +++ b/webrtc/video/video_send_stream.cc
|
| @@ -52,6 +52,7 @@
|
| 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 @@
|
| 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) {
|
| @@ -428,6 +430,7 @@
|
| &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_,
|
| @@ -885,7 +888,6 @@
|
| 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 =
|
|
|