New attempt to revert r2362, since drover failed.
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/640005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2368 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org
parent
cb89c6f914
commit
20e13edede
@ -1,5 +0,0 @@
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pwestin@webrtc.org
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stefan@webrtc.org
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henrik.lundin@webrtc.org
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mflodman@webrtc.org
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asapersson@webrtc.org
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@ -1,91 +0,0 @@
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/*
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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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#include "bitrate_estimator.h"
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namespace webrtc {
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enum { kBitrateAverageWindow = 2000 };
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BitRateStats::BitRateStats()
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:_dataSamples(), _accumulatedBytes(0)
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{
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}
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BitRateStats::~BitRateStats()
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{
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while (_dataSamples.size() > 0)
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{
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delete _dataSamples.front();
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_dataSamples.pop_front();
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}
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}
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void BitRateStats::Init()
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{
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_accumulatedBytes = 0;
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while (_dataSamples.size() > 0)
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{
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delete _dataSamples.front();
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_dataSamples.pop_front();
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}
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}
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void BitRateStats::Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs)
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{
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// Find an empty slot for storing the new sample and at the same time
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// accumulate the history.
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_dataSamples.push_back(new DataTimeSizeTuple(packetSizeBytes, nowMs));
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_accumulatedBytes += packetSizeBytes;
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EraseOld(nowMs);
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}
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void BitRateStats::EraseOld(WebRtc_Word64 nowMs)
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{
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while (_dataSamples.size() > 0)
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{
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if (nowMs - _dataSamples.front()->_timeCompleteMs >
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kBitrateAverageWindow)
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{
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// Delete old sample
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_accumulatedBytes -= _dataSamples.front()->_sizeBytes;
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delete _dataSamples.front();
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_dataSamples.pop_front();
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}
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else
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{
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break;
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}
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}
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}
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WebRtc_UWord32 BitRateStats::BitRate(WebRtc_Word64 nowMs)
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{
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// Calculate the average bit rate the past BITRATE_AVERAGE_WINDOW ms.
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// Removes any old samples from the list.
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EraseOld(nowMs);
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WebRtc_Word64 timeOldest = nowMs;
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if (_dataSamples.size() > 0)
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{
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timeOldest = _dataSamples.front()->_timeCompleteMs;
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}
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// Update average bit rate
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float denom = static_cast<float>(nowMs - timeOldest);
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if (nowMs == timeOldest)
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{
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// Calculate with a one second window when we haven't
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// received more than one packet.
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denom = 1000.0;
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}
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return static_cast<WebRtc_UWord32>(_accumulatedBytes * 8.0f * 1000.0f /
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denom + 0.5f);
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}
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} // namespace webrtc
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@ -1,50 +0,0 @@
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/*
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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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#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_
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#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_
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#include <list>
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#include "typedefs.h"
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namespace webrtc {
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class BitRateStats
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{
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public:
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BitRateStats();
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~BitRateStats();
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void Init();
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void Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs);
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WebRtc_UWord32 BitRate(WebRtc_Word64 nowMs);
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private:
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struct DataTimeSizeTuple
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{
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DataTimeSizeTuple(uint32_t sizeBytes, int64_t timeCompleteMs)
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:
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_sizeBytes(sizeBytes),
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_timeCompleteMs(timeCompleteMs) {}
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WebRtc_UWord32 _sizeBytes;
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WebRtc_Word64 _timeCompleteMs;
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};
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void EraseOld(WebRtc_Word64 nowMs);
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std::list<DataTimeSizeTuple*> _dataSamples;
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WebRtc_UWord32 _accumulatedBytes;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_
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@ -1,57 +0,0 @@
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/*
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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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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
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#include "typedefs.h"
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#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
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#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
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namespace webrtc {
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enum BandwidthUsage
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{
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kBwNormal,
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kBwOverusing,
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kBwUnderUsing
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};
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enum RateControlState
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{
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kRcHold,
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kRcIncrease,
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kRcDecrease
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};
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enum RateControlRegion
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{
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kRcNearMax,
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kRcAboveMax,
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kRcMaxUnknown
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};
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class RateControlInput
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{
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public:
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RateControlInput(BandwidthUsage bwState,
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WebRtc_UWord32 incomingBitRate,
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double noiseVar)
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: _bwState(bwState),
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_incomingBitRate(incomingBitRate),
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_noiseVar(noiseVar) {}
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BandwidthUsage _bwState;
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WebRtc_UWord32 _incomingBitRate;
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double _noiseVar;
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};
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} //namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
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@ -1,28 +0,0 @@
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/*
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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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#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
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#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
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#include <gmock/gmock.h>
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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namespace webrtc {
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class MockRemoteBitrateObserver : public RemoteBitrateObserver {
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public:
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MOCK_METHOD2(OnReceiveBitrateChanged,
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void(unsigned int ssrc, unsigned int bitrate));
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
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@ -1,82 +0,0 @@
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/*
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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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// RemoteBitrateEstimator
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// This class estimates the incoming bitrate capacity.
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#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
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#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
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#include <map>
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#include "modules/remote_bitrate_estimator/bitrate_estimator.h"
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#include "modules/remote_bitrate_estimator/overuse_detector.h"
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#include "modules/remote_bitrate_estimator/remote_rate_control.h"
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#include "system_wrappers/interface/critical_section_wrapper.h"
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#include "system_wrappers/interface/scoped_ptr.h"
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#include "typedefs.h"
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namespace webrtc {
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// RemoteBitrateObserver is used to signal changes in bitrate estimates for
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// the incoming stream.
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class RemoteBitrateObserver {
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public:
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// Called when a receive channel has a new bitrate estimate for the incoming
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// stream.
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virtual void OnReceiveBitrateChanged(unsigned int ssrc,
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unsigned int bitrate) = 0;
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virtual ~RemoteBitrateObserver() {}
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};
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class RemoteBitrateEstimator {
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public:
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explicit RemoteBitrateEstimator(RemoteBitrateObserver* observer);
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// Called for each incoming packet. If this is a new SSRC, a new
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// BitrateControl will be created.
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void IncomingPacket(unsigned int ssrc,
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int packet_size,
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int64_t arrival_time,
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uint32_t rtp_timestamp,
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int64_t packet_send_time);
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// Triggers a new estimate calculation for the stream identified by |ssrc|.
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void UpdateEstimate(unsigned int ssrc, int64_t time_now);
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// Set the current round-trip time experienced by the stream identified by
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// |ssrc|.
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void SetRtt(unsigned int ssrc);
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// Removes all data for |ssrc|.
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void RemoveStream(unsigned int ssrc);
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// Returns true if a valid estimate exists for a stream identified by |ssrc|
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// and sets |bitrate_bps| to the estimated bitrate in bits per second.
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bool LatestEstimate(unsigned int ssrc, unsigned int* bitrate_bps) const;
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private:
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struct BitrateControls {
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RemoteRateControl remote_rate;
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OverUseDetector overuse_detector;
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BitRateStats incoming_bitrate;
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};
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typedef std::map<unsigned int, BitrateControls> SsrcBitrateControlsMap;
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SsrcBitrateControlsMap bitrate_controls_;
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RemoteBitrateObserver* observer_;
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scoped_ptr<CriticalSectionWrapper> crit_sect_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
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@ -1,440 +0,0 @@
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/*
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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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#include <math.h>
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#include <stdlib.h> // abs
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#if _WIN32
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#include <windows.h>
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#endif
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#include "modules/remote_bitrate_estimator/overuse_detector.h"
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#include "modules/remote_bitrate_estimator/remote_rate_control.h"
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "system_wrappers/interface/trace.h"
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#ifdef WEBRTC_BWE_MATLAB
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extern MatlabEngine eng; // global variable defined elsewhere
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#endif
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#define INIT_CAPACITY_SLOPE 8.0/512.0
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#define DETECTOR_THRESHOLD 25.0
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#define OVER_USING_TIME_THRESHOLD 100
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#define MIN_FRAME_PERIOD_HISTORY_LEN 60
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namespace webrtc {
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OverUseDetector::OverUseDetector()
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: firstPacket_(true),
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currentFrame_(),
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prevFrame_(),
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numOfDeltas_(0),
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slope_(INIT_CAPACITY_SLOPE),
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offset_(0),
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E_(),
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processNoise_(),
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avgNoise_(0.0),
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varNoise_(500),
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threshold_(DETECTOR_THRESHOLD),
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tsDeltaHist_(),
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prevOffset_(0.0),
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timeOverUsing_(-1),
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overUseCounter_(0),
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#ifndef WEBRTC_BWE_MATLAB
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hypothesis_(kBwNormal) {
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#else
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plot1_(NULL),
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plot2_(NULL),
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plot3_(NULL),
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plot4_(NULL) {
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#endif
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E_[0][0] = 100;
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E_[1][1] = 1e-1;
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E_[0][1] = E_[1][0] = 0;
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processNoise_[0] = 1e-10;
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processNoise_[1] = 1e-2;
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}
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OverUseDetector::~OverUseDetector() {
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#ifdef WEBRTC_BWE_MATLAB
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if (plot1_) {
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eng.DeletePlot(plot1_);
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plot1_ = NULL;
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}
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if (plot2_) {
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eng.DeletePlot(plot2_);
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plot2_ = NULL;
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}
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if (plot3_) {
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eng.DeletePlot(plot3_);
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plot3_ = NULL;
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}
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if (plot4_) {
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eng.DeletePlot(plot4_);
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plot4_ = NULL;
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}
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#endif
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tsDeltaHist_.clear();
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}
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void OverUseDetector::Reset() {
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firstPacket_ = true;
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currentFrame_.size_ = 0;
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currentFrame_.completeTimeMs_ = -1;
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currentFrame_.timestamp_ = -1;
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prevFrame_.size_ = 0;
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prevFrame_.completeTimeMs_ = -1;
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prevFrame_.timestamp_ = -1;
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numOfDeltas_ = 0;
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slope_ = INIT_CAPACITY_SLOPE;
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offset_ = 0;
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E_[0][0] = 100;
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E_[1][1] = 1e-1;
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E_[0][1] = E_[1][0] = 0;
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processNoise_[0] = 1e-10;
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processNoise_[1] = 1e-2;
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avgNoise_ = 0.0;
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varNoise_ = 500;
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threshold_ = DETECTOR_THRESHOLD;
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prevOffset_ = 0.0;
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timeOverUsing_ = -1;
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overUseCounter_ = 0;
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hypothesis_ = kBwNormal;
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tsDeltaHist_.clear();
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}
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void OverUseDetector::Update(WebRtc_UWord16 packetSize,
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WebRtc_UWord32 timestamp,
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const WebRtc_Word64 nowMS) {
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#ifdef WEBRTC_BWE_MATLAB
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// Create plots
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const WebRtc_Word64 startTimeMs = nowMS;
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if (plot1_ == NULL) {
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plot1_ = eng.NewPlot(new MatlabPlot());
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plot1_->AddLine(1000, "b.", "scatter");
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}
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if (plot2_ == NULL) {
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plot2_ = eng.NewPlot(new MatlabPlot());
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plot2_->AddTimeLine(30, "b", "offset", startTimeMs);
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plot2_->AddTimeLine(30, "r--", "limitPos", startTimeMs);
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plot2_->AddTimeLine(30, "k.", "trigger", startTimeMs);
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plot2_->AddTimeLine(30, "ko", "detection", startTimeMs);
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// plot2_->AddTimeLine(30, "g", "slowMean", startTimeMs);
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}
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if (plot3_ == NULL) {
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plot3_ = eng.NewPlot(new MatlabPlot());
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plot3_->AddTimeLine(30, "b", "noiseVar", startTimeMs);
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}
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if (plot4_ == NULL) {
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plot4_ = eng.NewPlot(new MatlabPlot());
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// plot4_->AddTimeLine(60, "b", "p11", startTimeMs);
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// plot4_->AddTimeLine(60, "r", "p12", startTimeMs);
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plot4_->AddTimeLine(60, "g", "p22", startTimeMs);
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// plot4_->AddTimeLine(60, "g--", "p22_hat", startTimeMs);
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// plot4_->AddTimeLine(30, "b.-", "deltaFs", startTimeMs);
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}
|
||||
|
||||
#endif
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||||
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bool wrapped = false;
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bool completeFrame = false;
|
||||
if (currentFrame_.timestamp_ == -1) {
|
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currentFrame_.timestamp_ = timestamp;
|
||||
} else if (OldTimestamp(
|
||||
timestamp,
|
||||
static_cast<WebRtc_UWord32>(currentFrame_.timestamp_),
|
||||
&wrapped)) {
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||||
// Don't update with old data
|
||||
return;
|
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} else if (timestamp != currentFrame_.timestamp_) {
|
||||
// First packet of a later frame, the previous frame sample is ready
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"Frame complete at %I64i", currentFrame_.completeTimeMs_);
|
||||
if (prevFrame_.completeTimeMs_ >= 0) { // This is our second frame
|
||||
WebRtc_Word64 tDelta = 0;
|
||||
double tsDelta = 0;
|
||||
// Check for wrap
|
||||
OldTimestamp(
|
||||
static_cast<WebRtc_UWord32>(prevFrame_.timestamp_),
|
||||
static_cast<WebRtc_UWord32>(currentFrame_.timestamp_),
|
||||
&wrapped);
|
||||
CompensatedTimeDelta(currentFrame_, prevFrame_, tDelta, tsDelta,
|
||||
wrapped);
|
||||
UpdateKalman(tDelta, tsDelta, currentFrame_.size_,
|
||||
prevFrame_.size_);
|
||||
}
|
||||
// The new timestamp is now the current frame,
|
||||
// and the old timestamp becomes the previous frame.
|
||||
prevFrame_ = currentFrame_;
|
||||
currentFrame_.timestamp_ = timestamp;
|
||||
currentFrame_.size_ = 0;
|
||||
currentFrame_.completeTimeMs_ = -1;
|
||||
completeFrame = true;
|
||||
}
|
||||
// Accumulate the frame size
|
||||
currentFrame_.size_ += packetSize;
|
||||
currentFrame_.completeTimeMs_ = nowMS;
|
||||
}
|
||||
|
||||
BandwidthUsage OverUseDetector::State() const {
|
||||
return hypothesis_;
|
||||
}
|
||||
|
||||
double OverUseDetector::NoiseVar() const {
|
||||
return varNoise_;
|
||||
}
|
||||
|
||||
void OverUseDetector::SetRateControlRegion(RateControlRegion region) {
|
||||
switch (region) {
|
||||
case kRcMaxUnknown: {
|
||||
threshold_ = DETECTOR_THRESHOLD;
|
||||
break;
|
||||
}
|
||||
case kRcAboveMax:
|
||||
case kRcNearMax: {
|
||||
threshold_ = DETECTOR_THRESHOLD / 2;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void OverUseDetector::CompensatedTimeDelta(const FrameSample& currentFrame,
|
||||
const FrameSample& prevFrame,
|
||||
WebRtc_Word64& tDelta,
|
||||
double& tsDelta,
|
||||
bool wrapped) {
|
||||
numOfDeltas_++;
|
||||
if (numOfDeltas_ > 1000) {
|
||||
numOfDeltas_ = 1000;
|
||||
}
|
||||
// Add wrap-around compensation
|
||||
WebRtc_Word64 wrapCompensation = 0;
|
||||
if (wrapped) {
|
||||
wrapCompensation = static_cast<WebRtc_Word64>(1)<<32;
|
||||
}
|
||||
tsDelta = (currentFrame.timestamp_
|
||||
+ wrapCompensation
|
||||
- prevFrame.timestamp_) / 90.0;
|
||||
tDelta = currentFrame.completeTimeMs_ - prevFrame.completeTimeMs_;
|
||||
assert(tsDelta > 0);
|
||||
}
|
||||
|
||||
double OverUseDetector::CurrentDrift() {
|
||||
return 1.0;
|
||||
}
|
||||
|
||||
void OverUseDetector::UpdateKalman(WebRtc_Word64 tDelta,
|
||||
double tsDelta,
|
||||
WebRtc_UWord32 frameSize,
|
||||
WebRtc_UWord32 prevFrameSize) {
|
||||
const double minFramePeriod = UpdateMinFramePeriod(tsDelta);
|
||||
const double drift = CurrentDrift();
|
||||
// Compensate for drift
|
||||
const double tTsDelta = tDelta - tsDelta / drift;
|
||||
double fsDelta = static_cast<double>(frameSize) - prevFrameSize;
|
||||
|
||||
// Update the Kalman filter
|
||||
const double scaleFactor = minFramePeriod / (1000.0 / 30.0);
|
||||
E_[0][0] += processNoise_[0] * scaleFactor;
|
||||
E_[1][1] += processNoise_[1] * scaleFactor;
|
||||
|
||||
if ((hypothesis_ == kBwOverusing && offset_ < prevOffset_) ||
|
||||
(hypothesis_ == kBwUnderUsing && offset_ > prevOffset_)) {
|
||||
E_[1][1] += 10 * processNoise_[1] * scaleFactor;
|
||||
}
|
||||
|
||||
const double h[2] = {fsDelta, 1.0};
|
||||
const double Eh[2] = {E_[0][0]*h[0] + E_[0][1]*h[1],
|
||||
E_[1][0]*h[0] + E_[1][1]*h[1]};
|
||||
|
||||
const double residual = tTsDelta - slope_*h[0] - offset_;
|
||||
|
||||
const bool stableState =
|
||||
(BWE_MIN(numOfDeltas_, 60) * abs(offset_) < threshold_);
|
||||
// We try to filter out very late frames. For instance periodic key
|
||||
// frames doesn't fit the Gaussian model well.
|
||||
if (abs(residual) < 3 * sqrt(varNoise_)) {
|
||||
UpdateNoiseEstimate(residual, minFramePeriod, stableState);
|
||||
} else {
|
||||
UpdateNoiseEstimate(3 * sqrt(varNoise_), minFramePeriod, stableState);
|
||||
}
|
||||
|
||||
const double denom = varNoise_ + h[0]*Eh[0] + h[1]*Eh[1];
|
||||
|
||||
const double K[2] = {Eh[0] / denom,
|
||||
Eh[1] / denom};
|
||||
|
||||
const double IKh[2][2] = {{1.0 - K[0]*h[0], -K[0]*h[1]},
|
||||
{-K[1]*h[0], 1.0 - K[1]*h[1]}};
|
||||
const double e00 = E_[0][0];
|
||||
const double e01 = E_[0][1];
|
||||
|
||||
// Update state
|
||||
E_[0][0] = e00 * IKh[0][0] + E_[1][0] * IKh[0][1];
|
||||
E_[0][1] = e01 * IKh[0][0] + E_[1][1] * IKh[0][1];
|
||||
E_[1][0] = e00 * IKh[1][0] + E_[1][0] * IKh[1][1];
|
||||
E_[1][1] = e01 * IKh[1][0] + E_[1][1] * IKh[1][1];
|
||||
|
||||
// Covariance matrix, must be positive semi-definite
|
||||
assert(E_[0][0] + E_[1][1] >= 0 &&
|
||||
E_[0][0] * E_[1][1] - E_[0][1] * E_[1][0] >= 0 &&
|
||||
E_[0][0] >= 0);
|
||||
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
// plot4_->Append("p11",E_[0][0]);
|
||||
// plot4_->Append("p12",E_[0][1]);
|
||||
plot4_->Append("p22", E_[1][1]);
|
||||
// plot4_->Append("p22_hat", 0.5*(processNoise_[1] +
|
||||
// sqrt(processNoise_[1]*(processNoise_[1] + 4*varNoise_))));
|
||||
// plot4_->Append("deltaFs", fsDelta);
|
||||
plot4_->Plot();
|
||||
#endif
|
||||
slope_ = slope_ + K[0] * residual;
|
||||
prevOffset_ = offset_;
|
||||
offset_ = offset_ + K[1] * residual;
|
||||
|
||||
Detect(tsDelta);
|
||||
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
plot1_->Append("scatter",
|
||||
static_cast<double>(currentFrame_.size_) - prevFrame_.size_,
|
||||
static_cast<double>(tDelta-tsDelta));
|
||||
plot1_->MakeTrend("scatter", "slope", slope_, offset_, "k-");
|
||||
plot1_->MakeTrend("scatter", "thresholdPos",
|
||||
slope_, offset_ + 2 * sqrt(varNoise_), "r-");
|
||||
plot1_->MakeTrend("scatter", "thresholdNeg",
|
||||
slope_, offset_ - 2 * sqrt(varNoise_), "r-");
|
||||
plot1_->Plot();
|
||||
|
||||
plot2_->Append("offset", offset_);
|
||||
plot2_->Append("limitPos", threshold_/BWE_MIN(numOfDeltas_, 60));
|
||||
plot2_->Plot();
|
||||
|
||||
plot3_->Append("noiseVar", varNoise_);
|
||||
plot3_->Plot();
|
||||
#endif
|
||||
}
|
||||
|
||||
double OverUseDetector::UpdateMinFramePeriod(double tsDelta) {
|
||||
double minFramePeriod = tsDelta;
|
||||
if (tsDeltaHist_.size() >= MIN_FRAME_PERIOD_HISTORY_LEN) {
|
||||
std::list<double>::iterator firstItem = tsDeltaHist_.begin();
|
||||
tsDeltaHist_.erase(firstItem);
|
||||
}
|
||||
std::list<double>::iterator it = tsDeltaHist_.begin();
|
||||
for (; it != tsDeltaHist_.end(); it++) {
|
||||
minFramePeriod = BWE_MIN(*it, minFramePeriod);
|
||||
}
|
||||
tsDeltaHist_.push_back(tsDelta);
|
||||
return minFramePeriod;
|
||||
}
|
||||
|
||||
void OverUseDetector::UpdateNoiseEstimate(double residual,
|
||||
double tsDelta,
|
||||
bool stableState) {
|
||||
if (!stableState) {
|
||||
return;
|
||||
}
|
||||
// Faster filter during startup to faster adapt to the jitter level
|
||||
// of the network alpha is tuned for 30 frames per second, but
|
||||
double alpha = 0.01;
|
||||
if (numOfDeltas_ > 10*30) {
|
||||
alpha = 0.002;
|
||||
}
|
||||
// Only update the noise estimate if we're not over-using
|
||||
// beta is a function of alpha and the time delta since
|
||||
// the previous update.
|
||||
const double beta = pow(1 - alpha, tsDelta * 30.0 / 1000.0);
|
||||
avgNoise_ = beta * avgNoise_
|
||||
+ (1 - beta) * residual;
|
||||
varNoise_ = beta * varNoise_
|
||||
+ (1 - beta) * (avgNoise_ - residual) * (avgNoise_ - residual);
|
||||
if (varNoise_ < 1e-7) {
|
||||
varNoise_ = 1e-7;
|
||||
}
|
||||
}
|
||||
|
||||
BandwidthUsage OverUseDetector::Detect(double tsDelta) {
|
||||
if (numOfDeltas_ < 2) {
|
||||
return kBwNormal;
|
||||
}
|
||||
const double T = BWE_MIN(numOfDeltas_, 60) * offset_;
|
||||
if (abs(T) > threshold_) {
|
||||
if (offset_ > 0) {
|
||||
if (timeOverUsing_ == -1) {
|
||||
// Initialize the timer. Assume that we've been
|
||||
// over-using half of the time since the previous
|
||||
// sample.
|
||||
timeOverUsing_ = tsDelta / 2;
|
||||
} else {
|
||||
// Increment timer
|
||||
timeOverUsing_ += tsDelta;
|
||||
}
|
||||
overUseCounter_++;
|
||||
if (timeOverUsing_ > OVER_USING_TIME_THRESHOLD
|
||||
&& overUseCounter_ > 1) {
|
||||
if (offset_ >= prevOffset_) {
|
||||
#ifdef _DEBUG
|
||||
if (hypothesis_ != kBwOverusing) {
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwOverusing");
|
||||
}
|
||||
#endif
|
||||
timeOverUsing_ = 0;
|
||||
overUseCounter_ = 0;
|
||||
hypothesis_ = kBwOverusing;
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
plot2_->Append("detection", offset_); // plot it later
|
||||
#endif
|
||||
}
|
||||
}
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
plot2_->Append("trigger", offset_); // plot it later
|
||||
#endif
|
||||
} else {
|
||||
#ifdef _DEBUG
|
||||
if (hypothesis_ != kBwUnderUsing) {
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwUnderUsing");
|
||||
}
|
||||
#endif
|
||||
timeOverUsing_ = -1;
|
||||
overUseCounter_ = 0;
|
||||
hypothesis_ = kBwUnderUsing;
|
||||
}
|
||||
} else {
|
||||
#ifdef _DEBUG
|
||||
if (hypothesis_ != kBwNormal) {
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwNormal");
|
||||
}
|
||||
#endif
|
||||
timeOverUsing_ = -1;
|
||||
overUseCounter_ = 0;
|
||||
hypothesis_ = kBwNormal;
|
||||
}
|
||||
return hypothesis_;
|
||||
}
|
||||
|
||||
bool OverUseDetector::OldTimestamp(uint32_t newTimestamp,
|
||||
uint32_t existingTimestamp,
|
||||
bool* wrapped) {
|
||||
bool tmpWrapped =
|
||||
(newTimestamp < 0x0000ffff && existingTimestamp > 0xffff0000) ||
|
||||
(newTimestamp > 0xffff0000 && existingTimestamp < 0x0000ffff);
|
||||
*wrapped = tmpWrapped;
|
||||
if (existingTimestamp > newTimestamp && !tmpWrapped) {
|
||||
return true;
|
||||
} else if (existingTimestamp <= newTimestamp && !tmpWrapped) {
|
||||
return false;
|
||||
} else if (existingTimestamp < newTimestamp && tmpWrapped) {
|
||||
return true;
|
||||
} else {
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,91 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
|
||||
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
|
||||
|
||||
#include <list>
|
||||
|
||||
#include "modules/interface/module_common_types.h"
|
||||
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
|
||||
#include "typedefs.h" // NOLINT(build/include)
|
||||
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
#include "../test/BWEStandAlone/MatlabPlot.h"
|
||||
#endif
|
||||
|
||||
namespace webrtc {
|
||||
enum RateControlRegion;
|
||||
|
||||
class OverUseDetector {
|
||||
public:
|
||||
OverUseDetector();
|
||||
~OverUseDetector();
|
||||
void Update(const WebRtc_UWord16 packetSize,
|
||||
const WebRtc_UWord32 timestamp,
|
||||
const WebRtc_Word64 nowMS);
|
||||
BandwidthUsage State() const;
|
||||
void Reset();
|
||||
double NoiseVar() const;
|
||||
void SetRateControlRegion(RateControlRegion region);
|
||||
|
||||
private:
|
||||
struct FrameSample {
|
||||
FrameSample() : size_(0), completeTimeMs_(-1), timestamp_(-1) {}
|
||||
|
||||
WebRtc_UWord32 size_;
|
||||
WebRtc_Word64 completeTimeMs_;
|
||||
WebRtc_Word64 timestamp_;
|
||||
};
|
||||
|
||||
static bool OldTimestamp(uint32_t newTimestamp,
|
||||
uint32_t existingTimestamp,
|
||||
bool* wrapped);
|
||||
|
||||
void CompensatedTimeDelta(const FrameSample& currentFrame,
|
||||
const FrameSample& prevFrame,
|
||||
WebRtc_Word64& tDelta,
|
||||
double& tsDelta,
|
||||
bool wrapped);
|
||||
void UpdateKalman(WebRtc_Word64 tDelta,
|
||||
double tsDelta,
|
||||
WebRtc_UWord32 frameSize,
|
||||
WebRtc_UWord32 prevFrameSize);
|
||||
double UpdateMinFramePeriod(double tsDelta);
|
||||
void UpdateNoiseEstimate(double residual, double tsDelta, bool stableState);
|
||||
BandwidthUsage Detect(double tsDelta);
|
||||
double CurrentDrift();
|
||||
|
||||
bool firstPacket_;
|
||||
FrameSample currentFrame_;
|
||||
FrameSample prevFrame_;
|
||||
WebRtc_UWord16 numOfDeltas_;
|
||||
double slope_;
|
||||
double offset_;
|
||||
double E_[2][2];
|
||||
double processNoise_[2];
|
||||
double avgNoise_;
|
||||
double varNoise_;
|
||||
double threshold_;
|
||||
std::list<double> tsDeltaHist_;
|
||||
double prevOffset_;
|
||||
double timeOverUsing_;
|
||||
WebRtc_UWord16 overUseCounter_;
|
||||
BandwidthUsage hypothesis_;
|
||||
|
||||
#ifdef WEBRTC_BWE_MATLAB
|
||||
MatlabPlot* plot1_;
|
||||
MatlabPlot* plot2_;
|
||||
MatlabPlot* plot3_;
|
||||
MatlabPlot* plot4_;
|
||||
#endif
|
||||
};
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
|
||||
@ -1,102 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
||||
|
||||
#include "system_wrappers/interface/tick_util.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
RemoteBitrateEstimator::RemoteBitrateEstimator(
|
||||
RemoteBitrateObserver* observer)
|
||||
: observer_(observer),
|
||||
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
|
||||
assert(observer_);
|
||||
}
|
||||
|
||||
void RemoteBitrateEstimator::IncomingPacket(unsigned int ssrc,
|
||||
int packet_size,
|
||||
int64_t arrival_time,
|
||||
uint32_t rtp_timestamp,
|
||||
int64_t packet_send_time) {
|
||||
CriticalSectionScoped cs(crit_sect_.get());
|
||||
SsrcBitrateControlsMap::iterator it = bitrate_controls_.find(ssrc);
|
||||
if (it == bitrate_controls_.end()) {
|
||||
// This is a new SSRC. Adding to map.
|
||||
// TODO(holmer): If the channel changes SSRC the old SSRC will still be
|
||||
// around in this map until the channel is deleted. This is OK since the
|
||||
// callback will no longer be called for the old SSRC. This will be
|
||||
// automatically cleaned up when we have one RemoteBitrateEstimator per REMB
|
||||
// group.
|
||||
bitrate_controls_[ssrc] = BitrateControls();
|
||||
it = bitrate_controls_.find(ssrc);
|
||||
}
|
||||
OverUseDetector* overuse_detector =
|
||||
&bitrate_controls_[ssrc].overuse_detector;
|
||||
bitrate_controls_[ssrc].incoming_bitrate.Update(packet_size, arrival_time);
|
||||
const BandwidthUsage prior_state = overuse_detector->State();
|
||||
overuse_detector->Update(packet_size, rtp_timestamp, arrival_time);
|
||||
if (prior_state != overuse_detector->State() &&
|
||||
overuse_detector->State() == kBwOverusing) {
|
||||
// The first overuse should immediately trigger a new estimate.
|
||||
UpdateEstimate(ssrc, arrival_time);
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteBitrateEstimator::UpdateEstimate(unsigned int ssrc,
|
||||
int64_t time_now) {
|
||||
CriticalSectionScoped cs(crit_sect_.get());
|
||||
SsrcBitrateControlsMap::iterator it = bitrate_controls_.find(ssrc);
|
||||
if (it == bitrate_controls_.end()) {
|
||||
return;
|
||||
}
|
||||
OverUseDetector* overuse_detector = &it->second.overuse_detector;
|
||||
RemoteRateControl* remote_rate = &it->second.remote_rate;
|
||||
const RateControlInput input(overuse_detector->State(),
|
||||
it->second.incoming_bitrate.BitRate(time_now),
|
||||
overuse_detector->NoiseVar());
|
||||
const RateControlRegion region = remote_rate->Update(&input, time_now);
|
||||
unsigned int target_bitrate = remote_rate->UpdateBandwidthEstimate(time_now);
|
||||
if (remote_rate->ValidEstimate()) {
|
||||
observer_->OnReceiveBitrateChanged(ssrc, target_bitrate);
|
||||
}
|
||||
overuse_detector->SetRateControlRegion(region);
|
||||
}
|
||||
|
||||
void RemoteBitrateEstimator::SetRtt(unsigned int rtt) {
|
||||
CriticalSectionScoped cs(crit_sect_.get());
|
||||
for (SsrcBitrateControlsMap::iterator it = bitrate_controls_.begin();
|
||||
it != bitrate_controls_.end(); ++it) {
|
||||
it->second.remote_rate.SetRtt(rtt);
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteBitrateEstimator::RemoveStream(unsigned int ssrc) {
|
||||
CriticalSectionScoped cs(crit_sect_.get());
|
||||
// Ignoring the return value which is the number of elements erased.
|
||||
bitrate_controls_.erase(ssrc);
|
||||
}
|
||||
|
||||
bool RemoteBitrateEstimator::LatestEstimate(unsigned int ssrc,
|
||||
unsigned int* bitrate_bps) const {
|
||||
CriticalSectionScoped cs(crit_sect_.get());
|
||||
assert(bitrate_bps != NULL);
|
||||
SsrcBitrateControlsMap::const_iterator it = bitrate_controls_.find(ssrc);
|
||||
if (it == bitrate_controls_.end()) {
|
||||
return false;
|
||||
}
|
||||
if (!it->second.remote_rate.ValidEstimate()) {
|
||||
return false;
|
||||
}
|
||||
*bitrate_bps = it->second.remote_rate.LatestEstimate();
|
||||
return true;
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,71 +0,0 @@
|
||||
# Copyright (c) 2011 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.
|
||||
|
||||
{
|
||||
'targets': [
|
||||
{
|
||||
'target_name': 'remote_bitrate_estimator',
|
||||
'type': '<(library)',
|
||||
'dependencies': [
|
||||
# system_wrappers
|
||||
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
||||
],
|
||||
'include_dirs': [
|
||||
'include',
|
||||
'../rtp_rtcp/interface',
|
||||
'../interface',
|
||||
],
|
||||
'direct_dependent_settings': {
|
||||
'include_dirs': [
|
||||
'include',
|
||||
],
|
||||
},
|
||||
'sources': [
|
||||
# interface
|
||||
'include/bwe_defines.h',
|
||||
'include/remote_bitrate_estimator.h',
|
||||
|
||||
# source
|
||||
'bitrate_estimator.cc',
|
||||
'bitrate_estimator.h',
|
||||
'overuse_detector.cc',
|
||||
'overuse_detector.h',
|
||||
'remote_bitrate_estimator.cc',
|
||||
'remote_rate_control.cc',
|
||||
'remote_rate_control.h',
|
||||
], # source
|
||||
},
|
||||
], # targets
|
||||
'conditions': [
|
||||
['include_tests==1', {
|
||||
'targets': [
|
||||
{
|
||||
'target_name': 'remote_bitrate_estimator_unittests',
|
||||
'type': 'executable',
|
||||
'dependencies': [
|
||||
'remote_bitrate_estimator',
|
||||
'<(webrtc_root)/../testing/gmock.gyp:gmock',
|
||||
'<(webrtc_root)/../testing/gtest.gyp:gtest',
|
||||
'<(webrtc_root)/../test/test.gyp:test_support_main',
|
||||
],
|
||||
'sources': [
|
||||
'include/mock/mock_remote_bitrate_estimator.h',
|
||||
'bitrate_estimator_unittest.cc',
|
||||
'remote_bitrate_estimator_unittest.cc',
|
||||
],
|
||||
},
|
||||
], # targets
|
||||
}], # build_with_chromium
|
||||
], # conditions
|
||||
}
|
||||
|
||||
# Local Variables:
|
||||
# tab-width:2
|
||||
# indent-tabs-mode:nil
|
||||
# End:
|
||||
# vim: set expandtab tabstop=2 shiftwidth=2:
|
||||
@ -1,297 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
|
||||
// This file includes unit tests for RemoteBitrateEstimator.
|
||||
|
||||
#include <gtest/gtest.h>
|
||||
#include <list>
|
||||
|
||||
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
||||
#include "system_wrappers/interface/scoped_ptr.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
enum { kMtu = 1200 };
|
||||
|
||||
class TestBitrateObserver : public RemoteBitrateObserver {
|
||||
public:
|
||||
TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
|
||||
|
||||
void OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate) {
|
||||
latest_bitrate_ = bitrate;
|
||||
updated_ = true;
|
||||
}
|
||||
|
||||
bool updated() {
|
||||
bool updated = updated_;
|
||||
updated_ = false;
|
||||
return updated;
|
||||
}
|
||||
|
||||
unsigned int latest_bitrate() const {
|
||||
return latest_bitrate_;
|
||||
}
|
||||
|
||||
private:
|
||||
bool updated_;
|
||||
unsigned int latest_bitrate_;
|
||||
};
|
||||
|
||||
class StreamGenerator {
|
||||
public:
|
||||
struct Packet {
|
||||
int64_t send_time;
|
||||
int64_t arrival_time;
|
||||
uint32_t rtp_timestamp;
|
||||
unsigned int size;
|
||||
};
|
||||
|
||||
typedef std::list<Packet*> PacketList;
|
||||
|
||||
StreamGenerator(int fps, int bitrate_bps, int capacity, int64_t time_now)
|
||||
: fps_(fps),
|
||||
bitrate_bps_(bitrate_bps),
|
||||
capacity_(capacity),
|
||||
time_now_(time_now),
|
||||
prev_arrival_time_(time_now),
|
||||
rtp_timestamp_offset_(0xFFFFF000) {}
|
||||
|
||||
void SetCapacity(int capacity_bps) {
|
||||
ASSERT_GT(capacity_bps, 0);
|
||||
capacity_ = capacity_bps;
|
||||
}
|
||||
|
||||
void SetBitrate(int bitrate_bps) {
|
||||
ASSERT_GE(bitrate_bps, 0);
|
||||
bitrate_bps_ = bitrate_bps;
|
||||
}
|
||||
|
||||
void SetRtpTimestampOffset(uint32_t offset) {
|
||||
rtp_timestamp_offset_ = offset;
|
||||
}
|
||||
|
||||
void GenerateFrame(PacketList* packets) {
|
||||
ASSERT_FALSE(packets == NULL);
|
||||
ASSERT_TRUE(packets->empty());
|
||||
ASSERT_GT(fps_, 0);
|
||||
int bits_per_frame = bitrate_bps_ / fps_;
|
||||
int n_packets = std::max(bits_per_frame / (8 * kMtu), 1);
|
||||
int packet_size = bits_per_frame / (8 * n_packets);
|
||||
ASSERT_GE(n_packets, 0);
|
||||
for (int i = 0; i < n_packets; ++i) {
|
||||
Packet* packet = new Packet;
|
||||
packet->send_time = time_now_ + kSendSideOffsetMs;
|
||||
ASSERT_GT(capacity_, 0);
|
||||
packet->arrival_time = std::max(
|
||||
prev_arrival_time_ + 8 * 1000 * packet_size / capacity_,
|
||||
time_now_);
|
||||
packet->size = packet_size;
|
||||
packet->rtp_timestamp = rtp_timestamp_offset_ + 90 * packet->send_time;
|
||||
prev_arrival_time_ = packet->arrival_time;
|
||||
packets->push_back(packet);
|
||||
}
|
||||
time_now_ = time_now_ + 1000 / fps_;
|
||||
}
|
||||
|
||||
int64_t TimeNow() const {
|
||||
return time_now_;
|
||||
}
|
||||
|
||||
private:
|
||||
enum { kSendSideOffsetMs = 1000 };
|
||||
|
||||
int fps_;
|
||||
int bitrate_bps_;
|
||||
int capacity_;
|
||||
int64_t time_now_;
|
||||
int64_t prev_arrival_time_;
|
||||
uint32_t rtp_timestamp_offset_;
|
||||
};
|
||||
|
||||
class RemoteBitrateEstimatorTest : public ::testing::Test {
|
||||
protected:
|
||||
virtual void SetUp() {
|
||||
bitrate_observer_.reset(new TestBitrateObserver);
|
||||
bitrate_estimator_.reset(new RemoteBitrateEstimator(
|
||||
bitrate_observer_.get()));
|
||||
// Framerate: 30 fps; Start bitrate: 300 kbps; Link capacity: 1000 kbps,
|
||||
// Start time: 0.
|
||||
stream_generator_.reset(new StreamGenerator(30, 3e5, 1e6, 0));
|
||||
}
|
||||
|
||||
// Generates a frame of packets belonging to a stream at a given bitrate and
|
||||
// with a given ssrc. The stream is pushed through a very simple simulated
|
||||
// network, and is then given to the receive-side bandwidth estimator.
|
||||
void GenerateAndProcessFrame(unsigned int ssrc, unsigned int bitrate_bps) {
|
||||
stream_generator_->SetBitrate(bitrate_bps);
|
||||
StreamGenerator::PacketList packets;
|
||||
stream_generator_->GenerateFrame(&packets);
|
||||
int64_t last_arrival_time = -1;
|
||||
bool prev_was_decrease = false;
|
||||
while (!packets.empty()) {
|
||||
StreamGenerator::Packet* packet = packets.front();
|
||||
bitrate_estimator_->IncomingPacket(ssrc,
|
||||
packet->size,
|
||||
packet->arrival_time,
|
||||
packet->rtp_timestamp,
|
||||
-1);
|
||||
if (bitrate_observer_->updated()) {
|
||||
// Verify that new estimates only are triggered by an overuse and a
|
||||
// rate decrease.
|
||||
EXPECT_LE(bitrate_observer_->latest_bitrate(), bitrate_bps);
|
||||
EXPECT_FALSE(prev_was_decrease);
|
||||
prev_was_decrease = true;
|
||||
} else {
|
||||
prev_was_decrease = false;
|
||||
}
|
||||
last_arrival_time = packet->arrival_time;
|
||||
delete packet;
|
||||
packets.pop_front();
|
||||
}
|
||||
EXPECT_GT(last_arrival_time, -1);
|
||||
bitrate_estimator_->UpdateEstimate(ssrc, last_arrival_time);
|
||||
}
|
||||
|
||||
// Run the bandwidth estimator with a stream of |number_of_frames| frames.
|
||||
// Can for instance be used to run the estimator for some time to get it
|
||||
// into a steady state.
|
||||
unsigned int SteadyStateRun(unsigned int ssrc,
|
||||
int number_of_frames,
|
||||
unsigned int start_bitrate,
|
||||
unsigned int min_bitrate,
|
||||
unsigned int max_bitrate) {
|
||||
unsigned int bitrate_bps = start_bitrate;
|
||||
bool bitrate_update_seen = false;
|
||||
// Produce |number_of_frames| frames and give them to the estimator.
|
||||
for (int i = 0; i < number_of_frames; ++i) {
|
||||
GenerateAndProcessFrame(ssrc, bitrate_bps);
|
||||
if (bitrate_observer_->updated()) {
|
||||
EXPECT_LT(bitrate_observer_->latest_bitrate(), max_bitrate);
|
||||
EXPECT_GT(bitrate_observer_->latest_bitrate(), min_bitrate);
|
||||
bitrate_bps = bitrate_observer_->latest_bitrate();
|
||||
bitrate_update_seen = true;
|
||||
}
|
||||
}
|
||||
EXPECT_TRUE(bitrate_update_seen);
|
||||
return bitrate_bps;
|
||||
}
|
||||
|
||||
scoped_ptr<RemoteBitrateEstimator> bitrate_estimator_;
|
||||
scoped_ptr<TestBitrateObserver> bitrate_observer_;
|
||||
scoped_ptr<StreamGenerator> stream_generator_;
|
||||
};
|
||||
|
||||
TEST_F(RemoteBitrateEstimatorTest, TestInitialBehavior) {
|
||||
unsigned int bitrate_bps = 0;
|
||||
unsigned int ssrc = 0;
|
||||
int64_t time_now = 0;
|
||||
uint32_t timestamp = 0;
|
||||
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(ssrc, &bitrate_bps));
|
||||
bitrate_estimator_->UpdateEstimate(ssrc, time_now);
|
||||
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(ssrc, &bitrate_bps));
|
||||
EXPECT_FALSE(bitrate_observer_->updated());
|
||||
// Inserting a packet. Still no valid estimate. We need to wait 1 second.
|
||||
bitrate_estimator_->IncomingPacket(ssrc, kMtu, time_now,
|
||||
timestamp, -1);
|
||||
bitrate_estimator_->UpdateEstimate(ssrc, time_now);
|
||||
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(ssrc, &bitrate_bps));
|
||||
EXPECT_FALSE(bitrate_observer_->updated());
|
||||
// Waiting more than one second gives us a valid estimate.
|
||||
time_now += 1001;
|
||||
bitrate_estimator_->UpdateEstimate(ssrc, time_now);
|
||||
EXPECT_TRUE(bitrate_estimator_->LatestEstimate(ssrc, &bitrate_bps));
|
||||
EXPECT_EQ(bitrate_bps, 10734u);
|
||||
EXPECT_TRUE(bitrate_observer_->updated());
|
||||
EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps);
|
||||
}
|
||||
|
||||
// Make sure we initially increase the bitrate as expected.
|
||||
TEST_F(RemoteBitrateEstimatorTest, TestRateIncreaseRtpTimestamps) {
|
||||
const int kExpectedIterations = 323;
|
||||
unsigned int bitrate_bps = 30000;
|
||||
unsigned int ssrc = 0;
|
||||
int iterations = 0;
|
||||
// Feed the estimator with a stream of packets and verify that it reaches
|
||||
// 500 kbps at the expected time.
|
||||
while (bitrate_bps < 5e5) {
|
||||
GenerateAndProcessFrame(ssrc, bitrate_bps);
|
||||
if (bitrate_observer_->updated()) {
|
||||
EXPECT_GT(bitrate_observer_->latest_bitrate(), bitrate_bps);
|
||||
bitrate_bps = bitrate_observer_->latest_bitrate();
|
||||
}
|
||||
++iterations;
|
||||
ASSERT_LE(iterations, kExpectedIterations);
|
||||
}
|
||||
ASSERT_EQ(iterations, kExpectedIterations);
|
||||
}
|
||||
|
||||
// Verify that the time it takes for the estimator to reduce the bitrate when
|
||||
// the capacity is tightened stays the same.
|
||||
TEST_F(RemoteBitrateEstimatorTest, TestCapacityDropRtpTimestamps) {
|
||||
const unsigned int kSsrc = 0;
|
||||
const int kNumberOfFrames= 1000;
|
||||
const int kStartBitrate = 900e3;
|
||||
const int kMinExpectedBitrate = 800e3;
|
||||
const int kMaxExpectedBitrate = 1500e3;
|
||||
// Run in steady state to make the estimator converge.
|
||||
unsigned int bitrate_bps = SteadyStateRun(kSsrc, kNumberOfFrames,
|
||||
kStartBitrate, kMinExpectedBitrate,
|
||||
kMaxExpectedBitrate);
|
||||
// Reduce the capacity and verify the decrease time.
|
||||
stream_generator_->SetCapacity(500e3);
|
||||
int64_t bitrate_drop_time = 0;
|
||||
for (int i = 0; i < 1000; ++i) {
|
||||
GenerateAndProcessFrame(kSsrc, bitrate_bps);
|
||||
if (bitrate_observer_->updated()) {
|
||||
if (bitrate_observer_->latest_bitrate() <= 500e3) {
|
||||
bitrate_drop_time = stream_generator_->TimeNow();
|
||||
}
|
||||
bitrate_bps = bitrate_observer_->latest_bitrate();
|
||||
}
|
||||
}
|
||||
EXPECT_EQ(66000, bitrate_drop_time);
|
||||
}
|
||||
|
||||
// Verify that the time it takes for the estimator to reduce the bitrate when
|
||||
// the capacity is tightened stays the same. This test also verifies that we
|
||||
// handle wrap-arounds in this scenario.
|
||||
TEST_F(RemoteBitrateEstimatorTest, TestCapacityDropRtpTimestampsWrap) {
|
||||
const unsigned int kSsrc = 0;
|
||||
const int kFramerate= 30;
|
||||
const int kStartBitrate = 900e3;
|
||||
const int kMinExpectedBitrate = 800e3;
|
||||
const int kMaxExpectedBitrate = 1500e3;
|
||||
const int kSteadyStateTime = 5; // Seconds.
|
||||
// Trigger wrap right after the steady state run.
|
||||
stream_generator_->SetRtpTimestampOffset(
|
||||
std::numeric_limits<uint32_t>::max() - kSteadyStateTime * 90000);
|
||||
// Run in steady state to make the estimator converge.
|
||||
unsigned int bitrate_bps = SteadyStateRun(kSsrc,
|
||||
kSteadyStateTime * kFramerate,
|
||||
kStartBitrate,
|
||||
kMinExpectedBitrate,
|
||||
kMaxExpectedBitrate);
|
||||
// Reduce the capacity and verify the decrease time.
|
||||
stream_generator_->SetCapacity(500e3);
|
||||
int64_t bitrate_drop_time = 0;
|
||||
for (int i = 0; i < 1000; ++i) {
|
||||
GenerateAndProcessFrame(kSsrc, bitrate_bps);
|
||||
if (bitrate_observer_->updated()) {
|
||||
if (bitrate_observer_->latest_bitrate() <= 500e3) {
|
||||
bitrate_drop_time = stream_generator_->TimeNow();
|
||||
}
|
||||
bitrate_bps = bitrate_observer_->latest_bitrate();
|
||||
}
|
||||
}
|
||||
EXPECT_EQ(37356, bitrate_drop_time);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,489 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
|
||||
|
||||
#include <assert.h>
|
||||
#include <math.h>
|
||||
#include <string.h>
|
||||
#if _WIN32
|
||||
#include <windows.h>
|
||||
#endif
|
||||
|
||||
#include "system_wrappers/interface/trace.h"
|
||||
|
||||
#ifdef MATLAB
|
||||
extern MatlabEngine eng; // global variable defined elsewhere
|
||||
#endif
|
||||
|
||||
namespace webrtc {
|
||||
RemoteRateControl::RemoteRateControl()
|
||||
:
|
||||
_minConfiguredBitRate(30000),
|
||||
_maxConfiguredBitRate(30000000),
|
||||
_currentBitRate(_maxConfiguredBitRate),
|
||||
_maxHoldRate(0),
|
||||
_avgMaxBitRate(-1.0f),
|
||||
_varMaxBitRate(0.4f),
|
||||
_rcState(kRcHold),
|
||||
_cameFromState(kRcDecrease),
|
||||
_rcRegion(kRcMaxUnknown),
|
||||
_lastBitRateChange(-1),
|
||||
_currentInput(kBwNormal, 0, 1.0),
|
||||
_updated(false),
|
||||
_timeFirstIncomingEstimate(-1),
|
||||
_initializedBitRate(false),
|
||||
_avgChangePeriod(1000.0f),
|
||||
_lastChangeMs(-1),
|
||||
_beta(0.9f),
|
||||
_rtt(0)
|
||||
#ifdef MATLAB
|
||||
,_plot1(NULL),
|
||||
_plot2(NULL)
|
||||
#endif
|
||||
{
|
||||
}
|
||||
|
||||
RemoteRateControl::~RemoteRateControl()
|
||||
{
|
||||
#ifdef MATLAB
|
||||
eng.DeletePlot(_plot1);
|
||||
eng.DeletePlot(_plot2);
|
||||
#endif
|
||||
}
|
||||
|
||||
void RemoteRateControl::Reset()
|
||||
{
|
||||
_minConfiguredBitRate = 30000;
|
||||
_maxConfiguredBitRate = 30000000;
|
||||
_currentBitRate = _maxConfiguredBitRate;
|
||||
_maxHoldRate = 0;
|
||||
_avgMaxBitRate = -1.0f;
|
||||
_varMaxBitRate = 0.4f;
|
||||
_rcState = kRcHold;
|
||||
_cameFromState = kRcHold;
|
||||
_rcRegion = kRcMaxUnknown;
|
||||
_lastBitRateChange = -1;
|
||||
_avgChangePeriod = 1000.0f;
|
||||
_lastChangeMs = -1;
|
||||
_beta = 0.9f;
|
||||
_currentInput._bwState = kBwNormal;
|
||||
_currentInput._incomingBitRate = 0;
|
||||
_currentInput._noiseVar = 1.0;
|
||||
_updated = false;
|
||||
_timeFirstIncomingEstimate = -1;
|
||||
_initializedBitRate = false;
|
||||
}
|
||||
|
||||
bool RemoteRateControl::ValidEstimate() const {
|
||||
return _initializedBitRate;
|
||||
}
|
||||
|
||||
WebRtc_Word32 RemoteRateControl::SetConfiguredBitRates(
|
||||
WebRtc_UWord32 minBitRateBps, WebRtc_UWord32 maxBitRateBps)
|
||||
{
|
||||
if (minBitRateBps > maxBitRateBps)
|
||||
{
|
||||
return -1;
|
||||
}
|
||||
_minConfiguredBitRate = minBitRateBps;
|
||||
_maxConfiguredBitRate = maxBitRateBps;
|
||||
_currentBitRate = BWE_MIN(BWE_MAX(minBitRateBps, _currentBitRate),
|
||||
maxBitRateBps);
|
||||
return 0;
|
||||
}
|
||||
|
||||
WebRtc_UWord32 RemoteRateControl::LatestEstimate() const {
|
||||
return _currentBitRate;
|
||||
}
|
||||
|
||||
WebRtc_UWord32 RemoteRateControl::UpdateBandwidthEstimate(WebRtc_Word64 nowMS)
|
||||
{
|
||||
_currentBitRate = ChangeBitRate(_currentBitRate,
|
||||
_currentInput._incomingBitRate,
|
||||
_currentInput._noiseVar,
|
||||
nowMS);
|
||||
return _currentBitRate;
|
||||
}
|
||||
|
||||
void RemoteRateControl::SetRtt(unsigned int rtt) {
|
||||
_rtt = rtt;
|
||||
}
|
||||
|
||||
RateControlRegion RemoteRateControl::Update(const RateControlInput* input,
|
||||
WebRtc_Word64 nowMS)
|
||||
{
|
||||
assert(input);
|
||||
#ifdef MATLAB
|
||||
// Create plots
|
||||
if (_plot1 == NULL)
|
||||
{
|
||||
_plot1 = eng.NewPlot(new MatlabPlot());
|
||||
|
||||
_plot1->AddTimeLine(30, "b", "current");
|
||||
_plot1->AddTimeLine(30, "r-", "avgMax");
|
||||
_plot1->AddTimeLine(30, "r--", "pStdMax");
|
||||
_plot1->AddTimeLine(30, "r--", "nStdMax");
|
||||
_plot1->AddTimeLine(30, "r+", "max");
|
||||
_plot1->AddTimeLine(30, "g", "incoming");
|
||||
_plot1->AddTimeLine(30, "b+", "recovery");
|
||||
}
|
||||
if (_plot2 == NULL)
|
||||
{
|
||||
_plot2 = eng.NewPlot(new MatlabPlot());
|
||||
|
||||
_plot2->AddTimeLine(30, "b", "alpha");
|
||||
}
|
||||
#endif
|
||||
|
||||
// Set the initial bit rate value to what we're receiving the first second
|
||||
if (!_initializedBitRate)
|
||||
{
|
||||
if (_timeFirstIncomingEstimate < 0)
|
||||
{
|
||||
if (input->_incomingBitRate > 0)
|
||||
{
|
||||
_timeFirstIncomingEstimate = nowMS;
|
||||
}
|
||||
}
|
||||
else if (nowMS - _timeFirstIncomingEstimate > 1000 &&
|
||||
input->_incomingBitRate > 0)
|
||||
{
|
||||
_currentBitRate = input->_incomingBitRate;
|
||||
_initializedBitRate = true;
|
||||
}
|
||||
}
|
||||
|
||||
if (_updated && _currentInput._bwState == kBwOverusing)
|
||||
{
|
||||
// Only update delay factor and incoming bit rate. We always want to react on an over-use.
|
||||
_currentInput._noiseVar = input->_noiseVar;
|
||||
_currentInput._incomingBitRate = input->_incomingBitRate;
|
||||
return _rcRegion;
|
||||
}
|
||||
_updated = true;
|
||||
_currentInput = *input;
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Incoming rate = %u kbps", input->_incomingBitRate/1000);
|
||||
return _rcRegion;
|
||||
}
|
||||
|
||||
WebRtc_UWord32 RemoteRateControl::ChangeBitRate(WebRtc_UWord32 currentBitRate,
|
||||
WebRtc_UWord32 incomingBitRate,
|
||||
double noiseVar,
|
||||
WebRtc_Word64 nowMS)
|
||||
{
|
||||
if (!_updated)
|
||||
{
|
||||
return _currentBitRate;
|
||||
}
|
||||
_updated = false;
|
||||
UpdateChangePeriod(nowMS);
|
||||
ChangeState(_currentInput, nowMS);
|
||||
// calculated here because it's used in multiple places
|
||||
const float incomingBitRateKbps = incomingBitRate / 1000.0f;
|
||||
// Calculate the max bit rate std dev given the normalized
|
||||
// variance and the current incoming bit rate.
|
||||
const float stdMaxBitRate = sqrt(_varMaxBitRate * _avgMaxBitRate);
|
||||
bool recovery = false;
|
||||
switch (_rcState)
|
||||
{
|
||||
case kRcHold:
|
||||
{
|
||||
_maxHoldRate = BWE_MAX(_maxHoldRate, incomingBitRate);
|
||||
break;
|
||||
}
|
||||
case kRcIncrease:
|
||||
{
|
||||
if (_avgMaxBitRate >= 0)
|
||||
{
|
||||
if (incomingBitRateKbps > _avgMaxBitRate + 3 * stdMaxBitRate)
|
||||
{
|
||||
ChangeRegion(kRcMaxUnknown);
|
||||
_avgMaxBitRate = -1.0;
|
||||
}
|
||||
else if (incomingBitRateKbps > _avgMaxBitRate + 2.5 * stdMaxBitRate)
|
||||
{
|
||||
ChangeRegion(kRcAboveMax);
|
||||
}
|
||||
}
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"BWE: Response time: %f + %i + 10*33\n",
|
||||
_avgChangePeriod, _rtt);
|
||||
const WebRtc_UWord32 responseTime = static_cast<WebRtc_UWord32>(_avgChangePeriod + 0.5f) + _rtt + 300;
|
||||
double alpha = RateIncreaseFactor(nowMS, _lastBitRateChange,
|
||||
responseTime, noiseVar);
|
||||
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"BWE: _avgChangePeriod = %f ms; RTT = %u ms", _avgChangePeriod, _rtt);
|
||||
|
||||
currentBitRate = static_cast<WebRtc_UWord32>(currentBitRate * alpha) + 1000;
|
||||
if (_maxHoldRate > 0 && _beta * _maxHoldRate > currentBitRate)
|
||||
{
|
||||
currentBitRate = static_cast<WebRtc_UWord32>(_beta * _maxHoldRate);
|
||||
_avgMaxBitRate = _beta * _maxHoldRate / 1000.0f;
|
||||
ChangeRegion(kRcNearMax);
|
||||
recovery = true;
|
||||
#ifdef MATLAB
|
||||
_plot1->Append("recovery", _maxHoldRate/1000);
|
||||
#endif
|
||||
}
|
||||
_maxHoldRate = 0;
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"BWE: Increase rate to currentBitRate = %u kbps", currentBitRate/1000);
|
||||
_lastBitRateChange = nowMS;
|
||||
break;
|
||||
}
|
||||
case kRcDecrease:
|
||||
{
|
||||
if (incomingBitRate < _minConfiguredBitRate)
|
||||
{
|
||||
currentBitRate = _minConfiguredBitRate;
|
||||
}
|
||||
else
|
||||
{
|
||||
// Set bit rate to something slightly lower than max
|
||||
// to get rid of any self-induced delay.
|
||||
currentBitRate = static_cast<WebRtc_UWord32>(_beta * incomingBitRate + 0.5);
|
||||
if (currentBitRate > _currentBitRate)
|
||||
{
|
||||
// Avoid increasing the rate when over-using.
|
||||
if (_rcRegion != kRcMaxUnknown)
|
||||
{
|
||||
currentBitRate = static_cast<WebRtc_UWord32>(_beta * _avgMaxBitRate * 1000 + 0.5f);
|
||||
}
|
||||
currentBitRate = BWE_MIN(currentBitRate, _currentBitRate);
|
||||
}
|
||||
ChangeRegion(kRcNearMax);
|
||||
|
||||
if (incomingBitRateKbps < _avgMaxBitRate - 3 * stdMaxBitRate)
|
||||
{
|
||||
_avgMaxBitRate = -1.0f;
|
||||
}
|
||||
|
||||
UpdateMaxBitRateEstimate(incomingBitRateKbps);
|
||||
|
||||
#ifdef MATLAB
|
||||
_plot1->Append("max", incomingBitRateKbps);
|
||||
#endif
|
||||
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Decrease rate to currentBitRate = %u kbps", currentBitRate/1000);
|
||||
}
|
||||
// Stay on hold until the pipes are cleared.
|
||||
ChangeState(kRcHold);
|
||||
_lastBitRateChange = nowMS;
|
||||
break;
|
||||
}
|
||||
}
|
||||
if (!recovery && (incomingBitRate > 100000 || currentBitRate > 150000) &&
|
||||
currentBitRate > 1.5 * incomingBitRate)
|
||||
{
|
||||
// Allow changing the bit rate if we are operating at very low rates
|
||||
// Don't change the bit rate if the send side is too far off
|
||||
currentBitRate = _currentBitRate;
|
||||
_lastBitRateChange = nowMS;
|
||||
}
|
||||
#ifdef MATLAB
|
||||
if (_avgMaxBitRate >= 0.0f)
|
||||
{
|
||||
_plot1->Append("avgMax", _avgMaxBitRate);
|
||||
_plot1->Append("pStdMax", _avgMaxBitRate + 3*stdMaxBitRate);
|
||||
_plot1->Append("nStdMax", _avgMaxBitRate - 3*stdMaxBitRate);
|
||||
}
|
||||
_plot1->Append("incoming", incomingBitRate/1000);
|
||||
_plot1->Append("current", currentBitRate/1000);
|
||||
_plot1->Plot();
|
||||
#endif
|
||||
return currentBitRate;
|
||||
}
|
||||
|
||||
double RemoteRateControl::RateIncreaseFactor(WebRtc_Word64 nowMs, WebRtc_Word64 lastMs, WebRtc_UWord32 reactionTimeMs, double noiseVar) const
|
||||
{
|
||||
// alpha = 1.02 + B ./ (1 + exp(b*(tr - (c1*s2 + c2))))
|
||||
// Parameters
|
||||
const double B = 0.0407;
|
||||
const double b = 0.0025;
|
||||
const double c1 = -6700.0 / (33 * 33);
|
||||
const double c2 = 800.0;
|
||||
const double d = 0.85;
|
||||
|
||||
double alpha = 1.005 + B / (1 + exp( b * (d * reactionTimeMs - (c1 * noiseVar + c2))));
|
||||
|
||||
if (alpha < 1.005)
|
||||
{
|
||||
alpha = 1.005;
|
||||
}
|
||||
else if (alpha > 1.3)
|
||||
{
|
||||
alpha = 1.3;
|
||||
}
|
||||
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"BWE: alpha = %f", alpha);
|
||||
#ifdef MATLAB
|
||||
_plot2->Append("alpha", alpha);
|
||||
_plot2->Plot();
|
||||
#endif
|
||||
|
||||
if (lastMs > -1)
|
||||
{
|
||||
alpha = pow(alpha, (nowMs - lastMs) / 1000.0);
|
||||
}
|
||||
|
||||
if (_rcRegion == kRcNearMax)
|
||||
{
|
||||
// We're close to our previous maximum. Try to stabilize the
|
||||
// bit rate in this region, by increasing in smaller steps.
|
||||
alpha = alpha - (alpha - 1.0) / 2.0;
|
||||
}
|
||||
else if (_rcRegion == kRcMaxUnknown)
|
||||
{
|
||||
alpha = alpha + (alpha - 1.0) * 2.0;
|
||||
}
|
||||
|
||||
return alpha;
|
||||
}
|
||||
|
||||
void RemoteRateControl::UpdateChangePeriod(WebRtc_Word64 nowMs)
|
||||
{
|
||||
WebRtc_Word64 changePeriod = 0;
|
||||
if (_lastChangeMs > -1)
|
||||
{
|
||||
changePeriod = nowMs - _lastChangeMs;
|
||||
}
|
||||
_lastChangeMs = nowMs;
|
||||
_avgChangePeriod = 0.9f * _avgChangePeriod + 0.1f * changePeriod;
|
||||
}
|
||||
|
||||
void RemoteRateControl::UpdateMaxBitRateEstimate(float incomingBitRateKbps)
|
||||
{
|
||||
const float alpha = 0.05f;
|
||||
if (_avgMaxBitRate == -1.0f)
|
||||
{
|
||||
_avgMaxBitRate = incomingBitRateKbps;
|
||||
}
|
||||
else
|
||||
{
|
||||
_avgMaxBitRate = (1 - alpha) * _avgMaxBitRate +
|
||||
alpha * incomingBitRateKbps;
|
||||
}
|
||||
// Estimate the max bit rate variance and normalize the variance
|
||||
// with the average max bit rate.
|
||||
const float norm = BWE_MAX(_avgMaxBitRate, 1.0f);
|
||||
_varMaxBitRate = (1 - alpha) * _varMaxBitRate +
|
||||
alpha * (_avgMaxBitRate - incomingBitRateKbps) *
|
||||
(_avgMaxBitRate - incomingBitRateKbps) /
|
||||
norm;
|
||||
// 0.4 ~= 14 kbit/s at 500 kbit/s
|
||||
if (_varMaxBitRate < 0.4f)
|
||||
{
|
||||
_varMaxBitRate = 0.4f;
|
||||
}
|
||||
// 2.5f ~= 35 kbit/s at 500 kbit/s
|
||||
if (_varMaxBitRate > 2.5f)
|
||||
{
|
||||
_varMaxBitRate = 2.5f;
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteRateControl::ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs)
|
||||
{
|
||||
switch (_currentInput._bwState)
|
||||
{
|
||||
case kBwNormal:
|
||||
{
|
||||
if (_rcState == kRcHold)
|
||||
{
|
||||
_lastBitRateChange = nowMs;
|
||||
ChangeState(kRcIncrease);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case kBwOverusing:
|
||||
{
|
||||
if (_rcState != kRcDecrease)
|
||||
{
|
||||
ChangeState(kRcDecrease);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case kBwUnderUsing:
|
||||
{
|
||||
ChangeState(kRcHold);
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteRateControl::ChangeRegion(RateControlRegion region)
|
||||
{
|
||||
_rcRegion = region;
|
||||
switch (_rcRegion)
|
||||
{
|
||||
case kRcAboveMax:
|
||||
case kRcMaxUnknown:
|
||||
{
|
||||
_beta = 0.9f;
|
||||
break;
|
||||
}
|
||||
case kRcNearMax:
|
||||
{
|
||||
_beta = 0.95f;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteRateControl::ChangeState(RateControlState newState)
|
||||
{
|
||||
_cameFromState = _rcState;
|
||||
_rcState = newState;
|
||||
char state1[15];
|
||||
char state2[15];
|
||||
char state3[15];
|
||||
StateStr(_cameFromState, state1);
|
||||
StateStr(_rcState, state2);
|
||||
StateStr(_currentInput._bwState, state3);
|
||||
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
|
||||
"\t%s => %s due to %s\n", state1, state2, state3);
|
||||
}
|
||||
|
||||
void RemoteRateControl::StateStr(RateControlState state, char* str)
|
||||
{
|
||||
switch (state)
|
||||
{
|
||||
case kRcDecrease:
|
||||
strncpy(str, "DECREASE", 9);
|
||||
break;
|
||||
case kRcHold:
|
||||
strncpy(str, "HOLD", 5);
|
||||
break;
|
||||
case kRcIncrease:
|
||||
strncpy(str, "INCREASE", 9);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
void RemoteRateControl::StateStr(BandwidthUsage state, char* str)
|
||||
{
|
||||
switch (state)
|
||||
{
|
||||
case kBwNormal:
|
||||
strncpy(str, "NORMAL", 7);
|
||||
break;
|
||||
case kBwOverusing:
|
||||
strncpy(str, "OVER USING", 11);
|
||||
break;
|
||||
case kBwUnderUsing:
|
||||
strncpy(str, "UNDER USING", 12);
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1,83 +0,0 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
|
||||
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
|
||||
|
||||
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
|
||||
#include "typedefs.h"
|
||||
|
||||
#ifdef MATLAB
|
||||
#include "../test/BWEStandAlone/MatlabPlot.h"
|
||||
#endif
|
||||
|
||||
namespace webrtc {
|
||||
class RemoteRateControl
|
||||
{
|
||||
public:
|
||||
RemoteRateControl();
|
||||
~RemoteRateControl();
|
||||
WebRtc_Word32 SetConfiguredBitRates(WebRtc_UWord32 minBitRate,
|
||||
WebRtc_UWord32 maxBitRate);
|
||||
WebRtc_UWord32 LatestEstimate() const;
|
||||
WebRtc_UWord32 UpdateBandwidthEstimate(WebRtc_Word64 nowMS);
|
||||
void SetRtt(unsigned int rtt);
|
||||
RateControlRegion Update(const RateControlInput* input,
|
||||
WebRtc_Word64 nowMS);
|
||||
void Reset();
|
||||
|
||||
// Returns true if there is a valid estimate of the incoming bitrate, false
|
||||
// otherwise.
|
||||
bool ValidEstimate() const;
|
||||
|
||||
private:
|
||||
WebRtc_UWord32 ChangeBitRate(WebRtc_UWord32 currentBitRate,
|
||||
WebRtc_UWord32 incomingBitRate,
|
||||
double delayFactor,
|
||||
WebRtc_Word64 nowMS);
|
||||
double RateIncreaseFactor(WebRtc_Word64 nowMs,
|
||||
WebRtc_Word64 lastMs,
|
||||
WebRtc_UWord32 reactionTimeMs,
|
||||
double noiseVar) const;
|
||||
void UpdateChangePeriod(WebRtc_Word64 nowMs);
|
||||
void UpdateMaxBitRateEstimate(float incomingBitRateKbps);
|
||||
void ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs);
|
||||
void ChangeState(RateControlState newState);
|
||||
void ChangeRegion(RateControlRegion region);
|
||||
static void StateStr(RateControlState state, char* str);
|
||||
static void StateStr(BandwidthUsage state, char* str);
|
||||
|
||||
WebRtc_UWord32 _minConfiguredBitRate;
|
||||
WebRtc_UWord32 _maxConfiguredBitRate;
|
||||
WebRtc_UWord32 _currentBitRate;
|
||||
WebRtc_UWord32 _maxHoldRate;
|
||||
float _avgMaxBitRate;
|
||||
float _varMaxBitRate;
|
||||
RateControlState _rcState;
|
||||
RateControlState _cameFromState;
|
||||
RateControlRegion _rcRegion;
|
||||
WebRtc_Word64 _lastBitRateChange;
|
||||
RateControlInput _currentInput;
|
||||
bool _updated;
|
||||
WebRtc_Word64 _timeFirstIncomingEstimate;
|
||||
bool _initializedBitRate;
|
||||
|
||||
float _avgChangePeriod;
|
||||
WebRtc_Word64 _lastChangeMs;
|
||||
float _beta;
|
||||
unsigned int _rtt;
|
||||
#ifdef MATLAB
|
||||
MatlabPlot *_plot1;
|
||||
MatlabPlot *_plot2;
|
||||
#endif
|
||||
};
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
|
||||
35
src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
Normal file
35
src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
Normal file
@ -0,0 +1,35 @@
|
||||
# Copyright (c) 2011 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.
|
||||
|
||||
{
|
||||
'targets': [
|
||||
{
|
||||
'target_name': 'test_bwe',
|
||||
'type': 'executable',
|
||||
'dependencies': [
|
||||
'rtp_rtcp',
|
||||
'<(webrtc_root)/../test/test.gyp:test_support_main',
|
||||
'<(webrtc_root)/../testing/gtest.gyp:gtest',
|
||||
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
|
||||
],
|
||||
'include_dirs': [
|
||||
'../../source',
|
||||
],
|
||||
'sources': [
|
||||
'unit_test.cc',
|
||||
'../../source/bitrate.cc',
|
||||
],
|
||||
},
|
||||
],
|
||||
}
|
||||
|
||||
# Local Variables:
|
||||
# tab-width:2
|
||||
# indent-tabs-mode:nil
|
||||
# End:
|
||||
# vim: set expandtab tabstop=2 shiftwidth=2:
|
||||
@ -1,5 +1,5 @@
|
||||
/*
|
||||
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
||||
* Copyright (c) 2011 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
|
||||
@ -8,14 +8,15 @@
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
|
||||
/*
|
||||
* This file includes unit tests for the bitrate estimator.
|
||||
* This file includes unit tests for the bandwidth estimation and management
|
||||
*/
|
||||
|
||||
#include <gtest/gtest.h>
|
||||
|
||||
#include "typedefs.h"
|
||||
#include "bitrate_estimator.h"
|
||||
#include "Bitrate.h"
|
||||
|
||||
namespace {
|
||||
|
||||
@ -25,7 +26,7 @@ class BitRateStatsTest : public ::testing::Test
|
||||
{
|
||||
protected:
|
||||
BitRateStatsTest() {};
|
||||
BitRateStats bitRate;
|
||||
BitRateStats bitRate;
|
||||
};
|
||||
|
||||
TEST_F(BitRateStatsTest, TestStrictMode)
|
||||
@ -106,7 +106,7 @@ NORMAL_TESTS = {
|
||||
'rtp_rtcp_unittests': (True, True, True),
|
||||
'signal_processing_unittests': (True, True, True),
|
||||
'system_wrappers_unittests': (True, True, True),
|
||||
'remote_bitrate_estimator_unittests': (True, True, True),
|
||||
'test_bwe': (True, True, True),
|
||||
'test_fec': (True, True, True),
|
||||
'test_support_unittests': (True, True, True),
|
||||
'udp_transport_unittests': (True, True, True),
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user