Only use payload size within the receive/send interval for bitrate probing.

webrtc/modules/congestion_controller/probe_bitrate_estimator.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
 #define WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
 
+#include <limits>
 #include <map>
-#include <limits>
+#include <set>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 struct ProbingResult {
   static constexpr int kNoEstimate = -1;
 
   ProbingResult();
   ProbingResult(int bps, int64_t timestamp);
   bool valid() const;
 
   int bps;
   int64_t timestamp;
 };
 
 class ProbeBitrateEstimator {
  public:
   ProbeBitrateEstimator();
 
   // Should be called for every packet we receive feedback about. If the
   // packet was used for probing it will validate/calculate the resulting
   // bitrate and return the result.
   ProbingResult PacketFeedback(const PacketInfo& packet_info);
 
  private:
   struct AggregatedCluster {
-    int num_probes = 0;
+    std::set<uint16_t> sequence_numbers;
     int64_t first_send_ms = std::numeric_limits<int64_t>::max();
     int64_t last_send_ms = 0;
     int64_t first_receive_ms = std::numeric_limits<int64_t>::max();
     int64_t last_receive_ms = 0;
     size_t size = 0;
   };
 
   std::map<int, AggregatedCluster> clusters_;
   int last_valid_cluster_id_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_