07bc734459
- Move condition of 0 bps as max meaning 1gbps from SendSideBandwidthEstimation to BitrateController. - Remove condition on bitrate=0 meaning bandwidth estimation off as that could only happen when no observers existed and in which case the estimation would be ignored. - Add MaybeTriggerOnNetworkChanged which only runs rate allocation if any of the dependent variables has changed thus allowing to remove many of the bool returns that try to indicate if the estimation has changed which would not be aware if the observers have changed. - SendSideBandwidthEstimation now has a UpdateBitrate and has clear code paths to which calls update bitrate. - Changes in enforce_min_bitrate so the 10kbps min is set from the BitrateController and not from the outside this keep valid as observers are changed. R=henrik.lundin@webrtc.org, stefan@webrtc.org BUG=3065 Review URL: https://webrtc-codereview.appspot.com/10189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5752 4adac7df-926f-26a2-2b94-8c16560cd09d
61 lines
2.0 KiB
C++
61 lines
2.0 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 "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(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const;
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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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uint32_t rtt,
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int number_of_packets,
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uint32_t now_ms);
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void SetSendBitrate(uint32_t bitrate);
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void SetMinMaxBitrate(uint32_t min_bitrate, uint32_t max_bitrate);
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void SetMinBitrate(uint32_t min_bitrate);
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private:
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void UpdateEstimate(uint32_t now_ms);
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void CapBitrateToThresholds();
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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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uint8_t last_fraction_loss_;
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uint16_t last_round_trip_time_;
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uint32_t bwe_incoming_;
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uint32_t time_last_increase_;
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uint32_t time_last_decrease_;
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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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