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 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_ 106