/* * 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. */ #include "trace.h" #include "internal_defines.h" #include "jitter_estimator.h" #include "rtt_filter.h" #include "tick_time.h" #include #include #include namespace webrtc { VCMJitterEstimator::VCMJitterEstimator(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId) : _vcmId(vcmId), _receiverId(receiverId), _phi(0.97), _psi(0.9999), _alphaCountMax(400), _beta(0.9994), _thetaLow(0.000001), _nackLimit(3), // This should be 1 if the old // retransmition estimate is used. _nackWindowMS(15000), _numStdDevDelayOutlier(15), _numStdDevFrameSizeOutlier(3), _noiseStdDevs(2.33), // ~Less than 1% chance // (look up in normal distribution table)... _noiseStdDevOffset(30.0), // ...of getting 30 ms freezes _rttFilter(vcmId, receiverId) { Reset(); } VCMJitterEstimator& VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs) { if (this != &rhs) { memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov)); memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov)); _vcmId = rhs._vcmId; _receiverId = rhs._receiverId; _avgFrameSize = rhs._avgFrameSize; _varFrameSize = rhs._varFrameSize; _maxFrameSize = rhs._maxFrameSize; _fsSum = rhs._fsSum; _fsCount = rhs._fsCount; _lastUpdateT = rhs._lastUpdateT; _prevEstimate = rhs._prevEstimate; _prevFrameSize = rhs._prevFrameSize; _avgNoise = rhs._avgNoise; _alphaCount = rhs._alphaCount; _filterJitterEstimate = rhs._filterJitterEstimate; _startupCount = rhs._startupCount; _latestNackTimestamp = rhs._latestNackTimestamp; _nackCount = rhs._nackCount; _rttFilter = rhs._rttFilter; } return *this; } // Resets the JitterEstimate void VCMJitterEstimator::Reset() { _theta[0] = 1/(512e3/8); _theta[1] = 0; _varNoise = 4.0; _thetaCov[0][0] = 1e-4; _thetaCov[1][1] = 1e2; _thetaCov[0][1] = _thetaCov[1][0] = 0; _Qcov[0][0] = 2.5e-10; _Qcov[1][1] = 1e-10; _Qcov[0][1] = _Qcov[1][0] = 0; _avgFrameSize = 500; _maxFrameSize = 500; _varFrameSize = 100; _lastUpdateT = -1; _prevEstimate = -1.0; _prevFrameSize = 0; _avgNoise = 0.0; _alphaCount = 1; _filterJitterEstimate = 0.0; _latestNackTimestamp = 0; _nackCount = 0; _fsSum = 0; _fsCount = 0; _startupCount = 0; _rttFilter.Reset(); } void VCMJitterEstimator::ResetNackCount() { _nackCount = 0; } // Updates the estimates with the new measurements void VCMJitterEstimator::UpdateEstimate(WebRtc_Word64 frameDelayMS, WebRtc_UWord32 frameSizeBytes, bool incompleteFrame /* = false */) { WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "Jitter estimate updated with: frameSize=%d frameDelayMS=%d", frameSizeBytes, frameDelayMS); if (frameSizeBytes == 0) { return; } int deltaFS = frameSizeBytes - _prevFrameSize; if (_fsCount < kFsAccuStartupSamples) { _fsSum += frameSizeBytes; _fsCount++; } else if (_fsCount == kFsAccuStartupSamples) { // Give the frame size filter _avgFrameSize = static_cast(_fsSum) / static_cast(_fsCount); _fsCount++; } if (!incompleteFrame || frameSizeBytes > _avgFrameSize) { double avgFrameSize = _phi * _avgFrameSize + (1 - _phi) * frameSizeBytes; if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize)) { // Only update the average frame size if this sample wasn't a // key frame _avgFrameSize = avgFrameSize; } // Update the variance anyway since we want to capture cases where we only get // key frames. _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) * (frameSizeBytes - avgFrameSize) * (frameSizeBytes - avgFrameSize), 1.0); } // Update max frameSize estimate _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast(frameSizeBytes)); if (_prevFrameSize == 0) { _prevFrameSize = frameSizeBytes; return; } _prevFrameSize = frameSizeBytes; // Only update the Kalman filter if the sample is not considered // an extreme outlier. Even if it is an extreme outlier from a // delay point of view, if the frame size also is large the // deviation is probably due to an incorrect line slope. double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS); if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) || frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize)) { // Update the variance of the deviation from the // line given by the Kalman filter EstimateRandomJitter(deviation, incompleteFrame); // Prevent updating with frames which have been congested by a large // frame, and therefore arrives almost at the same time as that frame. // This can occur when we receive a large frame (key frame) which // has been delayed. The next frame is of normal size (delta frame), // and thus deltaFS will be << 0. This removes all frame samples // which arrives after a key frame. if ((!incompleteFrame || deviation >= 0.0) && static_cast(deltaFS) > - 0.25 * _maxFrameSize) { // Update the Kalman filter with the new data KalmanEstimateChannel(frameDelayMS, deltaFS); } } else { int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier; EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame); } // Post process the total estimated jitter if (_startupCount >= kStartupDelaySamples) { PostProcessEstimate(); } else { _startupCount++; } WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "Framesize statistics: max=%f average=%f", _maxFrameSize, _avgFrameSize); WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "The estimated slope is: theta=(%f, %f)", _theta[0], _theta[1]); WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "Random jitter: mean=%f variance=%f", _avgNoise, _varNoise); WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "Current jitter estimate: %f", _filterJitterEstimate); WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId), "Current max RTT: %u", _rttFilter.RttMs()); } // Updates the nack/packet ratio void VCMJitterEstimator::UpdateNackEstimate(bool retransmitted, WebRtc_Word64 /*wallClockMS = -1*/) { // Simplified since it seems to be hard to be sure if a // packet actually has been retransmitted or not, resulting // in a delay which varies up and down with one RTT. // The solution is to wait until _nackLimit retransmitts // has been received, then always add an RTT to the estimate. if (retransmitted && _nackCount < _nackLimit) { _nackCount++; } //if (wallClockMS == -1) //{ // wallClockMS = VCMTickTime::MillisecondTimestamp(); //} //if (retransmitted) //{ // if (_nackCount < _nackLimit) // { // _nackCount++; // } // _latestNackTimestamp = wallClockMS; //} //else if (_nackCount > 0 && wallClockMS - _latestNackTimestamp > _nackWindowMS) //{ // _nackCount = 0; //} } // Updates Kalman estimate of the channel // The caller is expected to sanity check the inputs. void VCMJitterEstimator::KalmanEstimateChannel(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes) { double Mh[2]; double hMh_sigma; double kalmanGain[2]; double measureRes; double t00, t01; // Kalman filtering // Prediction // M = M + Q _thetaCov[0][0] += _Qcov[0][0]; _thetaCov[0][1] += _Qcov[0][1]; _thetaCov[1][0] += _Qcov[1][0]; _thetaCov[1][1] += _Qcov[1][1]; // Kalman gain // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h') // h = [dFS 1] // Mh = M*h' // hMh_sigma = h*M*h' + R Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1]; Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1]; // sigma weights measurements with a small deltaFS as noisy and // measurements with large deltaFS as good if (_maxFrameSize < 1.0) { return; } double sigma = (300.0 * exp(-abs(static_cast(deltaFSBytes)) / (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise); if (sigma < 1.0) { sigma = 1.0; } hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma; if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0)) { assert(false); return; } kalmanGain[0] = Mh[0] / hMh_sigma; kalmanGain[1] = Mh[1] / hMh_sigma; // Correction // theta = theta + K*(dT - h*theta) measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]); _theta[0] += kalmanGain[0] * measureRes; _theta[1] += kalmanGain[1] * measureRes; if (_theta[0] < _thetaLow) { _theta[0] = _thetaLow; } // M = (I - K*h)*M t00 = _thetaCov[0][0]; t01 = _thetaCov[0][1]; _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 - kalmanGain[0] * _thetaCov[1][0]; _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 - kalmanGain[0] * _thetaCov[1][1]; _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) - kalmanGain[1] * deltaFSBytes * t00; _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) - kalmanGain[1] * deltaFSBytes * t01; // Covariance matrix, must be positive semi-definite assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 && _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 && _thetaCov[0][0] >= 0); } // Calculate difference in delay between a sample and the // expected delay estimated by the Kalman filter double VCMJitterEstimator::DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes) const { return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]); } // Estimates the random jitter by calculating the variance of the // sample distance from the line given by theta. void VCMJitterEstimator::EstimateRandomJitter(double d_dT, bool incompleteFrame) { double alpha; if (_alphaCount == 0) { assert(_alphaCount > 0); return; } alpha = static_cast(_alphaCount - 1) / static_cast(_alphaCount); _alphaCount++; if (_alphaCount > _alphaCountMax) { _alphaCount = _alphaCountMax; } double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT; double varNoise = alpha * _varNoise + (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise); if (!incompleteFrame || varNoise > _varNoise) { _avgNoise = avgNoise; _varNoise = varNoise; } if (_varNoise < 1.0) { // The variance should never be zero, since we might get // stuck and consider all samples as outliers. _varNoise = 1.0; } } double VCMJitterEstimator::NoiseThreshold() const { double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset; if (noiseThreshold < 1.0) { noiseThreshold = 1.0; } return noiseThreshold; } // Calculates the current jitter estimate from the filtered estimates double VCMJitterEstimator::CalculateEstimate() { double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold(); // A very low estimate (or negative) is neglected if (ret < 1.0) { if (_prevEstimate <= 0.01) { ret = 1.0; } else { ret = _prevEstimate; } } if (ret > 10000.0) // Sanity { ret = 10000.0; } _prevEstimate = ret; return ret; } void VCMJitterEstimator::PostProcessEstimate() { _filterJitterEstimate = CalculateEstimate(); } void VCMJitterEstimator::UpdateRtt(WebRtc_UWord32 rttMs) { _rttFilter.Update(rttMs); } void VCMJitterEstimator::UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes) { if (_maxFrameSize < frameSizeBytes) { _maxFrameSize = frameSizeBytes; } } // Returns the current filtered estimate if available, // otherwise tries to calculate an estimate. double VCMJitterEstimator::GetJitterEstimate() { double jitterMS = CalculateEstimate(); if (_filterJitterEstimate > jitterMS) { jitterMS = _filterJitterEstimate; } if (_nackCount >= _nackLimit) { return jitterMS + _rttFilter.RttMs(); } return jitterMS; } }