Move congestion controller processing to the pacer thread.

webrtc/modules/pacing/alr_detector.cc

 /*
  *  Copyright (c) 2016 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 18 matching lines @@
 }  // namespace
 
 namespace webrtc {
 
 AlrDetector::AlrDetector()
     : rate_(kMeasurementPeriodMs, RateStatistics::kBpsScale) {}
 
 AlrDetector::~AlrDetector() {}
 
 void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t now_ms) {
+  // TODO(nisse): It's unclear what guarantees there are that this
+  // function isn't called before SetEstimatedBitrate.

[stefan-webrtc, 2017/01/16 15:42:46]

I guess the guarantee is here in form of a DCHECK... :)

[nisse-webrtc, 2017/01/16 16:14:03]

Problem was, this thread reorg made this check crash. And then I fixed it by
adding an if (bytes_sent > 0) around the call.

Looks a bit brittle... So I guess the right thing is to tolerate zero
estimated_bitrate_bps_.

   RTC_DCHECK(estimated_bitrate_bps_);
 
   rate_.Update(bytes_sent, now_ms);
   rtc::Optional<uint32_t> rate = rate_.Rate(now_ms);
   if (!rate)
     return;
 
   int percentage = static_cast<int>(*rate) * 100 / estimated_bitrate_bps_;
   if (percentage < kAlrStartUsagePercent && !alr_started_time_ms_) {
     alr_started_time_ms_ = rtc::Optional<int64_t>(now_ms);
   } else if (percentage > kAlrEndUsagePercent && alr_started_time_ms_) {
     alr_started_time_ms_ = rtc::Optional<int64_t>();
   }
 }
 
 void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
   RTC_DCHECK(bitrate_bps);
   estimated_bitrate_bps_ = bitrate_bps;
 }
 
 rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
     const {
   return alr_started_time_ms_;
 }
 
 }  // namespace webrtc