17705a9c5a
git-svn-id: http://webrtc.googlecode.com/svn/trunk@74 4adac7df-926f-26a2-2b94-8c16560cd09d
161 lines
6.4 KiB
C++
161 lines
6.4 KiB
C++
/*
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* Copyright (c) 2011 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_VIDEO_CODING_JITTER_ESTIMATOR_H_
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#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
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#include "typedefs.h"
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#include "rtt_filter.h"
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namespace webrtc
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{
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class VCMJitterEstimator
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{
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public:
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VCMJitterEstimator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
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VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
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// Resets the estimate to the initial state
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void Reset();
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void ResetNackCount();
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// Updates the jitter estimate with the new data.
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//
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// Input:
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// - frameDelay : Delay-delta calculated by UTILDelayEstimate in milliseconds
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// - frameSize : Frame size of the current frame.
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// - incompleteFrame : Flags if the frame is used to update the estimate before it
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// was complete. Default is false.
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void UpdateEstimate(WebRtc_Word64 frameDelayMS,
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WebRtc_UWord32 frameSizeBytes,
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bool incompleteFrame = false);
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// Returns the current jitter estimate in milliseconds and adds
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// also adds an RTT dependent term in cases of retransmission.
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// Input:
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// - rttMultiplier : RTT param multiplier (when applicable).
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//
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// Return value : Jitter estimate in milliseconds
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double GetJitterEstimate(double rttMultiplier);
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// Updates the nack counter/timer.
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//
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// Input:
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// - retransmitted : True for a nacked frames, false otherwise
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// - wallClockMS : Used for testing
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void UpdateNackEstimate(bool retransmitted, WebRtc_Word64 wallClockMS = -1);
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// Updates the RTT filter.
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//
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// Input:
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// - rttMs : RTT in ms
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void UpdateRtt(WebRtc_UWord32 rttMs);
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void UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes);
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// A constant describing the delay from the jitter buffer
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// to the delay on the receiving side which is not accounted
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// for by the jitter buffer nor the decoding delay estimate.
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static const WebRtc_UWord32 OPERATING_SYSTEM_JITTER = 10;
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protected:
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// These are protected for better testing possibilities
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double _theta[2]; // Estimated line parameters (slope, offset)
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double _varNoise; // Variance of the time-deviation from the line
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private:
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// Updates the Kalman filter for the line describing
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// the frame size dependent jitter.
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//
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// Input:
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// - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
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// - deltaFSBytes : Frame size delta, i.e.
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// : frame size at time T minus frame size at time T-1
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void KalmanEstimateChannel(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes);
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// Updates the random jitter estimate, i.e. the variance
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// of the time deviations from the line given by the Kalman filter.
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//
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// Input:
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// - d_dT : The deviation from the kalman estimate
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// - incompleteFrame : True if the frame used to update the estimate
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// with was incomplete
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void EstimateRandomJitter(double d_dT, bool incompleteFrame);
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double NoiseThreshold() const;
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// Calculates the current jitter estimate.
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//
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// Return value : The current jitter estimate in milliseconds
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double CalculateEstimate();
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// Post process the calculated estimate
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void PostProcessEstimate();
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// Calculates the difference in delay between a sample and the
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// expected delay estimated by the Kalman filter.
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//
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// Input:
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// - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
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// - deltaFS : Frame size delta, i.e. frame size at time
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// T minus frame size at time T-1
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//
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// Return value : The difference in milliseconds
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double DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
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WebRtc_Word32 deltaFSBytes) const;
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// Constants, filter parameters
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WebRtc_Word32 _vcmId;
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WebRtc_Word32 _receiverId;
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const double _phi;
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const double _psi;
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const WebRtc_UWord32 _alphaCountMax;
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const double _beta;
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const double _thetaLow;
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const WebRtc_UWord32 _nackLimit;
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const WebRtc_UWord32 _nackWindowMS;
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const WebRtc_Word32 _numStdDevDelayOutlier;
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const WebRtc_Word32 _numStdDevFrameSizeOutlier;
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const double _noiseStdDevs;
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const double _noiseStdDevOffset;
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double _thetaCov[2][2]; // Estimate covariance
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double _Qcov[2][2]; // Process noise covariance
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double _avgFrameSize; // Average frame size
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double _varFrameSize; // Frame size variance
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double _maxFrameSize; // Largest frame size received (descending
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// with a factor _psi)
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WebRtc_UWord32 _fsSum;
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WebRtc_UWord32 _fsCount;
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WebRtc_Word64 _lastUpdateT;
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double _prevEstimate; // The previously returned jitter estimate
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WebRtc_UWord32 _prevFrameSize; // Frame size of the previous frame
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double _avgNoise; // Average of the random jitter
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WebRtc_UWord32 _alphaCount;
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double _filterJitterEstimate; // The filtered sum of jitter estimates
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WebRtc_UWord32 _startupCount;
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WebRtc_Word64 _latestNackTimestamp; // Timestamp in ms when the latest nack was seen
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WebRtc_UWord32 _nackCount; // Keeps track of the number of nacks received,
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// but never goes above _nackLimit
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VCMRttFilter _rttFilter;
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enum { kStartupDelaySamples = 30 };
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enum { kFsAccuStartupSamples = 5 };
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
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