44caf01c34
Modify bitrate controller to update bitrate based on process call and not only whenever a RTCP receiver block is received. Additionally: Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block. Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second). Did not touch decrease logic, however since it can be triggered more often it may decrease much faster and closer to the original written cap of once every 300ms + rtt. Note: rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap. bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block. BUG=3065 R=stefan@webrtc.org, mflodman@webrtc.org git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
72 lines
2.3 KiB
C++
72 lines
2.3 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(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const;
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// Call periodically to update estimate.
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void UpdateEstimate(uint32_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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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 CapBitrateToThresholds();
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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(uint32_t now_ms);
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std::deque<std::pair<uint32_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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uint32_t time_last_receiver_block_ms_;
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uint8_t last_fraction_loss_;
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uint16_t last_round_trip_time_ms_;
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uint32_t bwe_incoming_;
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uint32_t time_last_decrease_ms_;
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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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