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Side by Side Diff: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Undid unneccessary changes to rtp_rtcp module. Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 * FEC and NACK added bitrate is handled outside class 10 * FEC and NACK added bitrate is handled outside class
11 */ 11 */
12 12
13 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ 13 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
14 #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ 14 #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
15 15
16 #include <deque> 16 #include <deque>
17 #include <utility> 17 #include <utility>
18 #include <vector> 18 #include <vector>
19 19
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 class RtcEventLog; 25 class RtcEventLog;
26 26
27 class SendSideBandwidthEstimation { 27 class SendSideBandwidthEstimation {
28 public: 28 public:
29 SendSideBandwidthEstimation(); 29 SendSideBandwidthEstimation(RtcEventLog* event_log);
the sun 2016/03/21 13:03:09 explicit
ivoc 2016/03/22 13:44:55 Thanks, done.
30 virtual ~SendSideBandwidthEstimation(); 30 virtual ~SendSideBandwidthEstimation();
31 31
32 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; 32 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
33 33
34 // Call periodically to update estimate. 34 // Call periodically to update estimate.
35 void UpdateEstimate(int64_t now_ms); 35 void UpdateEstimate(int64_t now_ms);
36 36
37 // Call when we receive a RTCP message with TMMBR or REMB. 37 // Call when we receive a RTCP message with TMMBR or REMB.
38 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); 38 void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
39 39
40 // Call when a new delay-based estimate is available. 40 // Call when a new delay-based estimate is available.
41 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps); 41 void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);
42 42
43 // Call when we receive a RTCP message with a ReceiveBlock. 43 // Call when we receive a RTCP message with a ReceiveBlock.
44 void UpdateReceiverBlock(uint8_t fraction_loss, 44 void UpdateReceiverBlock(uint8_t fraction_loss,
45 int64_t rtt, 45 int64_t rtt,
46 int number_of_packets, 46 int number_of_packets,
47 int64_t now_ms); 47 int64_t now_ms);
48 48
49 void SetSendBitrate(int bitrate); 49 void SetSendBitrate(int bitrate);
50 void SetMinMaxBitrate(int min_bitrate, int max_bitrate); 50 void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
51 int GetMinBitrate() const; 51 int GetMinBitrate() const;
52 52
53 void SetEventLog(RtcEventLog* event_log);
54
55 private: 53 private:
56 enum UmaState { kNoUpdate, kFirstDone, kDone }; 54 enum UmaState { kNoUpdate, kFirstDone, kDone };
57 55
58 bool IsInStartPhase(int64_t now_ms) const; 56 bool IsInStartPhase(int64_t now_ms) const;
59 57
60 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); 58 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
61 59
62 // Returns the input bitrate capped to the thresholds defined by the max, 60 // Returns the input bitrate capped to the thresholds defined by the max,
63 // min and incoming bandwidth. 61 // min and incoming bandwidth.
64 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); 62 uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate);
(...skipping 20 matching lines...) Expand all
85 int64_t last_round_trip_time_ms_; 83 int64_t last_round_trip_time_ms_;
86 84
87 uint32_t bwe_incoming_; 85 uint32_t bwe_incoming_;
88 uint32_t delay_based_bitrate_bps_; 86 uint32_t delay_based_bitrate_bps_;
89 int64_t time_last_decrease_ms_; 87 int64_t time_last_decrease_ms_;
90 int64_t first_report_time_ms_; 88 int64_t first_report_time_ms_;
91 int initially_lost_packets_; 89 int initially_lost_packets_;
92 int bitrate_at_2_seconds_kbps_; 90 int bitrate_at_2_seconds_kbps_;
93 UmaState uma_update_state_; 91 UmaState uma_update_state_;
94 std::vector<bool> rampup_uma_stats_updated_; 92 std::vector<bool> rampup_uma_stats_updated_;
95 RtcEventLog* event_log_; 93 RtcEventLog* event_log_;
the sun 2016/03/21 13:03:09 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
ivoc 2016/03/22 13:44:55 Done.
96 }; 94 };
97 } // namespace webrtc 95 } // namespace webrtc
98 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ 96 #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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