| /* |
| * 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); |
| ~ProbeBitrateEstimator(); |
| |
| // 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); |
| |
| rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps(); |
| |
| 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_; |
| rtc::Optional<int> estimated_bitrate_bps_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_ |