Fix for alr detection bug on pause.

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 388 matching lines @@
     int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
     if (ret > 0 || (ret == 0 && !probing_send_failure_))
       return ret;
   }
   return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
 }
 
 void PacedSender::Process() {
   int64_t now_us = clock_->TimeInMicroseconds();
   rtc::CritScope cs(&critsect_);
-  int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
+  int64_t elapsed_time_ms = std::min(
+      kMaxIntervalTimeMs, (now_us - time_last_update_us_ + 500) / 1000);
   int target_bitrate_kbps = pacing_bitrate_kbps_;
 
   if (paused_) {
     PacedPacketInfo pacing_info;
     time_last_update_us_ = now_us;
     // We can not send padding unless a normal packet has first been sent. If we
     // do, timestamps get messed up.
     if (packet_counter_ == 0)
       return;
     size_t bytes_sent = SendPadding(1, pacing_info);
-    alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
+    alr_detector_->OnBytesSent(bytes_sent, elapsed_time_ms);
     return;
   }
 
   if (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, 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);
   }
 
   time_last_update_us_ = now_us;
 
   bool is_probing = prober_->IsProbing();
   PacedPacketInfo pacing_info;
   size_t bytes_sent = 0;
   size_t recommended_probe_size = 0;
   if (is_probing) {
@@ skipping 103 matching lines @@
   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