| /* |
| * 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 "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| |
| namespace webrtc { |
| class SendSideBandwidthEstimation { |
| public: |
| SendSideBandwidthEstimation(); |
| virtual ~SendSideBandwidthEstimation(); |
| |
| // Call when we receive a RTCP message with TMMBR or REMB |
| // Return true if new_bitrate is valid. |
| bool UpdateBandwidthEstimate(const uint32_t bandwidth, |
| uint32_t* new_bitrate, |
| uint8_t* fraction_lost, |
| uint16_t* rtt); |
| |
| // Call when we receive a RTCP message with a ReceiveBlock |
| // Return true if new_bitrate is valid. |
| bool UpdatePacketLoss(const int number_of_packets, |
| const uint32_t rtt, |
| const uint32_t now_ms, |
| uint8_t* loss, |
| uint32_t* new_bitrate); |
| |
| // Return false if no bandwidth estimate is available |
| bool AvailableBandwidth(uint32_t* bandwidth) const; |
| void SetSendBitrate(const uint32_t bitrate); |
| void SetMinMaxBitrate(const uint32_t min_bitrate, const uint32_t max_bitrate); |
| |
| private: |
| bool ShapeSimple(const uint8_t loss, const uint32_t rtt, |
| const uint32_t now_ms, uint32_t* bitrate); |
| |
| uint32_t CalcTFRCbps(uint16_t rtt, uint8_t loss); |
| |
| enum { kBWEIncreaseIntervalMs = 1000 }; |
| enum { kBWEDecreaseIntervalMs = 300 }; |
| enum { kLimitNumPackets = 20 }; |
| enum { kAvgPacketSizeBytes = 1000 }; |
| |
| CriticalSectionWrapper* critsect_; |
| |
| // incoming filters |
| int accumulate_lost_packets_Q8_; |
| int accumulate_expected_packets_; |
| |
| uint32_t bitrate_; |
| uint32_t min_bitrate_configured_; |
| uint32_t max_bitrate_configured_; |
| |
| uint8_t last_fraction_loss_; |
| uint16_t last_round_trip_time_; |
| |
| uint32_t bwe_incoming_; |
| uint32_t time_last_increase_; |
| uint32_t time_last_decrease_; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |