|  | /* | 
|  | *  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 <map> | 
|  | #include <limits> | 
|  |  | 
|  | #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | class RtcEventLog; | 
|  |  | 
|  | class ProbeBitrateEstimator { | 
|  | public: | 
|  | explicit ProbeBitrateEstimator(RtcEventLog* event_log); | 
|  |  | 
|  | // Should be called for every probe packet we receive feedback about. | 
|  | // Returns the estimated bitrate if the probe completes a valid cluster. | 
|  | int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback); | 
|  |  | 
|  | private: | 
|  | struct AggregatedCluster { | 
|  | int num_probes = 0; | 
|  | 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; | 
|  | int size_last_send = 0; | 
|  | int size_first_receive = 0; | 
|  | int size_total = 0; | 
|  | }; | 
|  |  | 
|  | // Erases old cluster data that was seen before |timestamp_ms|. | 
|  | void EraseOldClusters(int64_t timestamp_ms); | 
|  |  | 
|  | std::map<int, AggregatedCluster> clusters_; | 
|  | RtcEventLog* const event_log_; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_ |