Line data Source code
1 : /*
2 : * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 : *
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
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : *
10 : * FEC and NACK added bitrate is handled outside class
11 : */
12 :
13 : #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
14 : #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
15 :
16 : #include <deque>
17 : #include <utility>
18 : #include <vector>
19 :
20 : #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
21 : #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
22 :
23 : namespace webrtc {
24 :
25 : class RtcEventLog;
26 :
27 0 : class SendSideBandwidthEstimation {
28 : public:
29 : SendSideBandwidthEstimation() = delete;
30 : explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
31 : virtual ~SendSideBandwidthEstimation();
32 :
33 : void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
34 :
35 : // Call periodically to update estimate.
36 : void UpdateEstimate(int64_t now_ms);
37 :
38 : // Call when we receive a RTCP message with TMMBR or REMB.
39 : void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
40 :
41 : // Call when a new delay-based estimate is available.
42 : void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);
43 :
44 : // Call when we receive a RTCP message with a ReceiveBlock.
45 : void UpdateReceiverBlock(uint8_t fraction_loss,
46 : int64_t rtt,
47 : int number_of_packets,
48 : int64_t now_ms);
49 :
50 : void SetBitrates(int send_bitrate,
51 : int min_bitrate,
52 : int max_bitrate);
53 : void SetSendBitrate(int bitrate);
54 : void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
55 : int GetMinBitrate() const;
56 :
57 : private:
58 : enum UmaState { kNoUpdate, kFirstDone, kDone };
59 :
60 : bool IsInStartPhase(int64_t now_ms) const;
61 :
62 : void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
63 :
64 : // Returns the input bitrate capped to the thresholds defined by the max,
65 : // min and incoming bandwidth.
66 : uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate);
67 :
68 : // Updates history of min bitrates.
69 : // After this method returns min_bitrate_history_.front().second contains the
70 : // min bitrate used during last kBweIncreaseIntervalMs.
71 : void UpdateMinHistory(int64_t now_ms);
72 :
73 : std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
74 :
75 : // incoming filters
76 : int lost_packets_since_last_loss_update_Q8_;
77 : int expected_packets_since_last_loss_update_;
78 :
79 : uint32_t bitrate_;
80 : uint32_t min_bitrate_configured_;
81 : uint32_t max_bitrate_configured_;
82 : int64_t last_low_bitrate_log_ms_;
83 :
84 : bool has_decreased_since_last_fraction_loss_;
85 : int64_t last_feedback_ms_;
86 : int64_t last_packet_report_ms_;
87 : int64_t last_timeout_ms_;
88 : uint8_t last_fraction_loss_;
89 : uint8_t last_logged_fraction_loss_;
90 : int64_t last_round_trip_time_ms_;
91 :
92 : uint32_t bwe_incoming_;
93 : uint32_t delay_based_bitrate_bps_;
94 : int64_t time_last_decrease_ms_;
95 : int64_t first_report_time_ms_;
96 : int initially_lost_packets_;
97 : int bitrate_at_2_seconds_kbps_;
98 : UmaState uma_update_state_;
99 : std::vector<bool> rampup_uma_stats_updated_;
100 : RtcEventLog* event_log_;
101 : int64_t last_rtc_event_log_ms_;
102 : bool in_timeout_experiment_;
103 : };
104 : } // namespace webrtc
105 : #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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