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Side by Side Diff: webrtc/modules/congestion_controller/probe_bitrate_estimator.h

Issue 2997883002: Video/Screenshare loopback tool.
Patch Set: Rebase Created 3 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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21 21
22 class ProbeBitrateEstimator { 22 class ProbeBitrateEstimator {
23 public: 23 public:
24 explicit ProbeBitrateEstimator(RtcEventLog* event_log); 24 explicit ProbeBitrateEstimator(RtcEventLog* event_log);
25 ~ProbeBitrateEstimator(); 25 ~ProbeBitrateEstimator();
26 26
27 // Should be called for every probe packet we receive feedback about. 27 // Should be called for every probe packet we receive feedback about.
28 // Returns the estimated bitrate if the probe completes a valid cluster. 28 // Returns the estimated bitrate if the probe completes a valid cluster.
29 int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback); 29 int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback);
30 30
31 rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps(); 31 rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps(int64_t timeout = 0);
32 32
33 private: 33 private:
34 struct AggregatedCluster { 34 struct AggregatedCluster {
35 int num_probes = 0; 35 int num_probes = 0;
36 int64_t first_send_ms = std::numeric_limits<int64_t>::max(); 36 int64_t first_send_ms = std::numeric_limits<int64_t>::max();
37 int64_t last_send_ms = 0; 37 int64_t last_send_ms = 0;
38 int64_t first_receive_ms = std::numeric_limits<int64_t>::max(); 38 int64_t first_receive_ms = std::numeric_limits<int64_t>::max();
39 int64_t last_receive_ms = 0; 39 int64_t last_receive_ms = 0;
40 int size_last_send = 0; 40 int size_last_send = 0;
41 int size_first_receive = 0; 41 int size_first_receive = 0;
42 int size_total = 0; 42 int size_total = 0;
43 }; 43 };
44 44
45 // Erases old cluster data that was seen before |timestamp_ms|. 45 // Erases old cluster data that was seen before |timestamp_ms|.
46 void EraseOldClusters(int64_t timestamp_ms); 46 void EraseOldClusters(int64_t timestamp_ms);
47 47
48 std::map<int, AggregatedCluster> clusters_; 48 std::map<int, AggregatedCluster> clusters_;
49 RtcEventLog* const event_log_; 49 RtcEventLog* const event_log_;
50 rtc::Optional<int> estimated_bitrate_bps_; 50 rtc::Optional<int> estimated_bitrate_bps_;
51 int64_t last_estimate_update_ms_ = 0;
52 int64_t start_ = 0;
53 int cluster_id_ = -1;
51 }; 54 };
52 55
53 } // namespace webrtc 56 } // namespace webrtc
54 57
55 #endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_ 58 #endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
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