| /* |
| * Copyright (c) 2012 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. |
| * |
| * FEC and NACK added bitrate is handled outside class |
| */ |
| |
| #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
| #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
| |
| #include <deque> |
| #include <utility> |
| #include <vector> |
| |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
| |
| namespace webrtc { |
| |
| class RtcEventLog; |
| |
| class SendSideBandwidthEstimation { |
| public: |
| SendSideBandwidthEstimation(); |
| virtual ~SendSideBandwidthEstimation(); |
| |
| void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const; |
| |
| // Call periodically to update estimate. |
| void UpdateEstimate(int64_t now_ms); |
| |
| // Call when we receive a RTCP message with TMMBR or REMB. |
| void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth); |
| |
| // Call when a new delay-based estimate is available. |
| void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps); |
| |
| // Call when we receive a RTCP message with a ReceiveBlock. |
| void UpdateReceiverBlock(uint8_t fraction_loss, |
| int64_t rtt, |
| int number_of_packets, |
| int64_t now_ms); |
| |
| void SetBitrates(int send_bitrate, |
| int min_bitrate, |
| int max_bitrate); |
| void SetSendBitrate(int bitrate); |
| void SetMinMaxBitrate(int min_bitrate, int max_bitrate); |
| int GetMinBitrate() const; |
| |
| void SetEventLog(RtcEventLog* event_log); |
| |
| private: |
| enum UmaState { kNoUpdate, kFirstDone, kDone }; |
| |
| bool IsInStartPhase(int64_t now_ms) const; |
| |
| void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets); |
| |
| // Returns the input bitrate capped to the thresholds defined by the max, |
| // min and incoming bandwidth. |
| uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate); |
| |
| // Updates history of min bitrates. |
| // After this method returns min_bitrate_history_.front().second contains the |
| // min bitrate used during last kBweIncreaseIntervalMs. |
| void UpdateMinHistory(int64_t now_ms); |
| |
| std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_; |
| |
| // incoming filters |
| int lost_packets_since_last_loss_update_Q8_; |
| int expected_packets_since_last_loss_update_; |
| |
| uint32_t bitrate_; |
| uint32_t min_bitrate_configured_; |
| uint32_t max_bitrate_configured_; |
| int64_t last_low_bitrate_log_ms_; |
| |
| bool has_decreased_since_last_fraction_loss_; |
| int64_t time_last_receiver_block_ms_; |
| uint8_t last_fraction_loss_; |
| int64_t last_round_trip_time_ms_; |
| |
| uint32_t bwe_incoming_; |
| uint32_t delay_based_bitrate_bps_; |
| int64_t time_last_decrease_ms_; |
| int64_t first_report_time_ms_; |
| int initially_lost_packets_; |
| int bitrate_at_2_seconds_kbps_; |
| UmaState uma_update_state_; |
| std::vector<bool> rampup_uma_stats_updated_; |
| RtcEventLog* event_log_; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |