Wire up experiment for improved screenshare bwe.

webrtc/modules/pacing/paced_sender.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 246 matching lines @@
       padding_budget_(new paced_sender::IntervalBudget(0)),
       prober_(new BitrateProber(event_log)),
       probing_send_failure_(false),
       estimated_bitrate_bps_(0),
       min_send_bitrate_kbps_(0u),
       max_padding_bitrate_kbps_(0u),
       pacing_bitrate_kbps_(0),
       time_last_update_us_(clock->TimeInMicroseconds()),
       first_sent_packet_ms_(-1),
       packets_(new paced_sender::PacketQueue(clock)),
-      packet_counter_(0) {
+      packet_counter_(0),
+      pacing_factor_(kDefaultPaceMultiplier),
+      queue_time_limit(kMaxQueueLengthMs) {
   UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
 }
 
 PacedSender::~PacedSender() {}
 
 void PacedSender::CreateProbeCluster(int bitrate_bps) {
   rtc::CritScope cs(&critsect_);
   prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
 }
 
@@ skipping 29 matching lines @@
 
 void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
   if (bitrate_bps == 0)
     LOG(LS_ERROR) << "PacedSender is not designed to handle 0 bitrate.";
   rtc::CritScope cs(&critsect_);
   estimated_bitrate_bps_ = bitrate_bps;
   padding_budget_->set_target_rate_kbps(
       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
   pacing_bitrate_kbps_ =
       std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
-      kDefaultPaceMultiplier;
+      pacing_factor_;
   alr_detector_->SetEstimatedBitrate(bitrate_bps);
 }
 
 void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
                                        int padding_bitrate) {
   rtc::CritScope cs(&critsect_);
   min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
   pacing_bitrate_kbps_ =
       std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
-      kDefaultPaceMultiplier;
+      pacing_factor_;
   max_padding_bitrate_kbps_ = padding_bitrate / 1000;
   padding_budget_->set_target_rate_kbps(
       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
 }
 
 void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                uint32_t ssrc,
                                uint16_t sequence_number,
                                int64_t capture_time_ms,
                                size_t bytes,
@@ skipping 74 matching lines @@
   time_last_update_us_ = now_us;
   int target_bitrate_kbps = pacing_bitrate_kbps_;
   if (!paused_ && elapsed_time_ms > 0) {
     size_t queue_size_bytes = packets_->SizeInBytes();
     if (queue_size_bytes > 0) {
       // Assuming equal size packets and input/output rate, the average packet
       // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
       // time constraint shall be met. Determine bitrate needed for that.
       packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
       int64_t avg_time_left_ms = std::max<int64_t>(
-          1, kMaxQueueLengthMs - packets_->AverageQueueTimeMs());
+          1, queue_time_limit - packets_->AverageQueueTimeMs());
       int min_bitrate_needed_kbps =
           static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
       if (min_bitrate_needed_kbps > target_bitrate_kbps)
         target_bitrate_kbps = min_bitrate_needed_kbps;
     }
 
     media_budget_->set_target_rate_kbps(target_bitrate_kbps);
 
     elapsed_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms);
     UpdateBudgetWithElapsedTime(elapsed_time_ms);
@@ skipping 95 matching lines @@
 
 void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
   media_budget_->IncreaseBudget(delta_time_ms);
   padding_budget_->IncreaseBudget(delta_time_ms);
 }
 
 void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
   media_budget_->UseBudget(bytes_sent);
   padding_budget_->UseBudget(bytes_sent);
 }
+
+void PacedSender::SetPacingFactor(float pacing_factor) {
+  rtc::CritScope cs(&critsect_);
+  pacing_factor_ = pacing_factor;
+}
+
+void PacedSender::SetQueueTimeLimit(int limit_ms) {
+  rtc::CritScope cs(&critsect_);
+  queue_time_limit = limit_ms;
+}
+
 }  // namespace webrtc