Relanding "Pass time constant to bwe smoothing filter."

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 315 matching lines @@
 
   void SetTransportOverhead(int transport_overhead_per_packet);
 
  private:
   class CheckEncoderActivityTask;
   class EncoderReconfiguredTask;
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
-                            int64_t rtt) override;
+                            int64_t rtt,
+                            int64_t probing_interval_ms) override;
 
   // Implements webrtc::VCMProtectionCallback.
   int ProtectionRequest(const FecProtectionParams* delta_params,
                         const FecProtectionParams* key_params,
                         uint32_t* sent_video_rate_bps,
                         uint32_t* sent_nack_rate_bps,
                         uint32_t* sent_fec_rate_bps) override;
 
   // Implements OverheadObserver.
   void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
@@ skipping 816 matching lines @@
 void VideoSendStreamImpl::SignalNetworkState(NetworkState state) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_->rtp.rtcp_mode
                                                 : RtcpMode::kOff);
   }
 }
 
 uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
                                                uint8_t fraction_loss,
-                                               int64_t rtt) {
+                                               int64_t rtt,
+                                               int64_t probing_interval_ms) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   RTC_DCHECK(payload_router_.active())
       << "VideoSendStream::Start has not been called.";
 
   if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
       "Enabled") {
     // Substract overhead from bitrate.
     rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
     int packets_per_second =
         std::ceil(bitrate_bps / (8 * (config_->rtp.max_packet_size +
@@ skipping 75 matching lines @@
   const uint16_t mtu = static_cast<uint16_t>(
       config_->rtp.max_packet_size + transport_overhead_bytes_per_packet);
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetTransportOverhead(transport_overhead_bytes_per_packet);
     rtp_rtcp->SetMaxTransferUnit(mtu);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc