Add logging of delay-based bandwidth estimate.

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 2015 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/modules/congestion_controller/transport_feedback_adapter.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
 const int64_t kSendTimeHistoryWindowMs = 60000;
 const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
 
 class PacketInfoComparator {
  public:
   inline bool operator()(const PacketInfo& lhs, const PacketInfo& rhs) {
     if (lhs.arrival_time_ms != rhs.arrival_time_ms)
       return lhs.arrival_time_ms < rhs.arrival_time_ms;
     if (lhs.send_time_ms != rhs.send_time_ms)
       return lhs.send_time_ms < rhs.send_time_ms;
     return lhs.sequence_number < rhs.sequence_number;
   }
 };
 
 TransportFeedbackAdapter::TransportFeedbackAdapter(
+    RtcEventLog* event_log,
     Clock* clock,
     BitrateController* bitrate_controller)
     : send_side_bwe_with_overhead_(webrtc::field_trial::FindFullName(
                                        "WebRTC-SendSideBwe-WithOverhead") ==
                                    "Enabled"),
       transport_overhead_bytes_per_packet_(0),
       send_time_history_(clock, kSendTimeHistoryWindowMs),
+      event_log_(event_log),
       clock_(clock),
       current_offset_ms_(kNoTimestamp),
       last_timestamp_us_(kNoTimestamp),
       bitrate_controller_(bitrate_controller) {}
 
 TransportFeedbackAdapter::~TransportFeedbackAdapter() {}
 
 void TransportFeedbackAdapter::InitBwe() {
   rtc::CritScope cs(&bwe_lock_);
-  delay_based_bwe_.reset(new DelayBasedBwe(clock_));
+  delay_based_bwe_.reset(new DelayBasedBwe(event_log_, clock_));
 }
 
 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
                                          size_t length,
                                          int probe_cluster_id) {
   rtc::CritScope cs(&lock_);
   if (send_side_bwe_with_overhead_) {
     length += transport_overhead_bytes_per_packet_;
   }
   send_time_history_.AddAndRemoveOld(sequence_number, length, probe_cluster_id);
@@ skipping 97 matching lines @@
   return last_packet_feedback_vector_;
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc