e590416722
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out. BUG=4323 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45549004 Cr-Commit-Position: refs/heads/master@{#8864}
85 lines
2.8 KiB
C++
85 lines
2.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*
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* FEC and NACK added bitrate is handled outside class
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*/
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#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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#define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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#include <deque>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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class SendSideBandwidthEstimation {
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public:
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SendSideBandwidthEstimation();
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virtual ~SendSideBandwidthEstimation();
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void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
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// Call periodically to update estimate.
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void UpdateEstimate(int64_t now_ms);
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// Call when we receive a RTCP message with TMMBR or REMB.
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void UpdateReceiverEstimate(uint32_t bandwidth);
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// Call when we receive a RTCP message with a ReceiveBlock.
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void UpdateReceiverBlock(uint8_t fraction_loss,
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int64_t rtt,
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int number_of_packets,
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int64_t now_ms);
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void SetSendBitrate(int bitrate);
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void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
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int GetMinBitrate() const;
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private:
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enum UmaState { kNoUpdate, kFirstDone, kDone };
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bool IsInStartPhase(int64_t now_ms) const;
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void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
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// Returns the input bitrate capped to the thresholds defined by the max,
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// min and incoming bandwidth.
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uint32_t CapBitrateToThresholds(uint32_t bitrate);
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// Updates history of min bitrates.
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// After this method returns min_bitrate_history_.front().second contains the
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// min bitrate used during last kBweIncreaseIntervalMs.
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void UpdateMinHistory(int64_t now_ms);
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std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
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// incoming filters
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int accumulate_lost_packets_Q8_;
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int accumulate_expected_packets_;
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uint32_t bitrate_;
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uint32_t min_bitrate_configured_;
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uint32_t max_bitrate_configured_;
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int64_t time_last_receiver_block_ms_;
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uint8_t last_fraction_loss_;
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int64_t last_round_trip_time_ms_;
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uint32_t bwe_incoming_;
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int64_t time_last_decrease_ms_;
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int64_t first_report_time_ms_;
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int initially_lost_packets_;
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int bitrate_at_2_seconds_kbps_;
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UmaState uma_update_state_;
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std::vector<bool> rampup_uma_stats_updated_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
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