webrtc/modules/video_coding/main/source/jitter_estimator.h
mikhal@google.com 17705a9c5a Review URL: http://webrtc-codereview.appspot.com/28004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@74 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-14 17:54:20 +00:00

161 lines
6.4 KiB
C++

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