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New attempt to revert r2362, since drover failed.

Browse Source 
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 2012-06-05 13:07:56 +00:00
parent cb89c6f914
commit 20e13edede
16 changed files with 41 additions and 1891 deletions
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5
src/modules/remote_bitrate_estimator/OWNERS
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pwestin@webrtc.org
stefan@webrtc.org
henrik.lundin@webrtc.org
mflodman@webrtc.org
asapersson@webrtc.org

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src/modules/remote_bitrate_estimator/bitrate_estimator.cc
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/*
* 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 "bitrate_estimator.h"
namespace webrtc {
enum { kBitrateAverageWindow = 2000 };
BitRateStats::BitRateStats()
:_dataSamples(), _accumulatedBytes(0)
{
}
BitRateStats::~BitRateStats()
{
while (_dataSamples.size() > 0)
{
delete _dataSamples.front();
_dataSamples.pop_front();
}
}
void BitRateStats::Init()
{
_accumulatedBytes = 0;
while (_dataSamples.size() > 0)
{
delete _dataSamples.front();
_dataSamples.pop_front();
}
}
void BitRateStats::Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs)
{
// Find an empty slot for storing the new sample and at the same time
// accumulate the history.
_dataSamples.push_back(new DataTimeSizeTuple(packetSizeBytes, nowMs));
_accumulatedBytes += packetSizeBytes;
EraseOld(nowMs);
}
void BitRateStats::EraseOld(WebRtc_Word64 nowMs)
{
while (_dataSamples.size() > 0)
{
if (nowMs - _dataSamples.front()->_timeCompleteMs >
kBitrateAverageWindow)
{
// Delete old sample
_accumulatedBytes -= _dataSamples.front()->_sizeBytes;
delete _dataSamples.front();
_dataSamples.pop_front();
}
else
{
break;
}
}
}
WebRtc_UWord32 BitRateStats::BitRate(WebRtc_Word64 nowMs)
{
// Calculate the average bit rate the past BITRATE_AVERAGE_WINDOW ms.
// Removes any old samples from the list.
EraseOld(nowMs);
WebRtc_Word64 timeOldest = nowMs;
if (_dataSamples.size() > 0)
{
timeOldest = _dataSamples.front()->_timeCompleteMs;
}
// Update average bit rate
float denom = static_cast<float>(nowMs - timeOldest);
if (nowMs == timeOldest)
{
// Calculate with a one second window when we haven't
// received more than one packet.
denom = 1000.0;
}
return static_cast<WebRtc_UWord32>(_accumulatedBytes * 8.0f * 1000.0f /
denom + 0.5f);
}
} // namespace webrtc

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src/modules/remote_bitrate_estimator/bitrate_estimator.h
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/*
* 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_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_
#include <list>
#include "typedefs.h"
namespace webrtc {
class BitRateStats
{
public:
BitRateStats();
~BitRateStats();
void Init();
void Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs);
WebRtc_UWord32 BitRate(WebRtc_Word64 nowMs);
private:
struct DataTimeSizeTuple
{
DataTimeSizeTuple(uint32_t sizeBytes, int64_t timeCompleteMs)
:
_sizeBytes(sizeBytes),
_timeCompleteMs(timeCompleteMs) {}
WebRtc_UWord32 _sizeBytes;
WebRtc_Word64 _timeCompleteMs;
};
void EraseOld(WebRtc_Word64 nowMs);
std::list<DataTimeSizeTuple*> _dataSamples;
WebRtc_UWord32 _accumulatedBytes;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_BITRATE_ESTIMATOR_H_

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src/modules/remote_bitrate_estimator/include/bwe_defines.h
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/*
* 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_BWE_DEFINES_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
#include "typedefs.h"
#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
namespace webrtc {
enum BandwidthUsage
{
kBwNormal,
kBwOverusing,
kBwUnderUsing
};
enum RateControlState
{
kRcHold,
kRcIncrease,
kRcDecrease
};
enum RateControlRegion
{
kRcNearMax,
kRcAboveMax,
kRcMaxUnknown
};
class RateControlInput
{
public:
RateControlInput(BandwidthUsage bwState,
WebRtc_UWord32 incomingBitRate,
double noiseVar)
: _bwState(bwState),
_incomingBitRate(incomingBitRate),
_noiseVar(noiseVar) {}
BandwidthUsage _bwState;
WebRtc_UWord32 _incomingBitRate;
double _noiseVar;
};
} //namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_

28
src/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h
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/*
* 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_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_
#include <gmock/gmock.h>
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
namespace webrtc {
class MockRemoteBitrateObserver : public RemoteBitrateObserver {
public:
MOCK_METHOD2(OnReceiveBitrateChanged,
void(unsigned int ssrc, unsigned int bitrate));
};
} // namespace webrtc
#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_MOCK_MOCK_REMOTE_BITRATE_ESTIMATOR_H_

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src/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
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/*
* 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.
*/
// RemoteBitrateEstimator
// This class estimates the incoming bitrate capacity.
#ifndef WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
#include <map>
#include "modules/remote_bitrate_estimator/bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "typedefs.h"
namespace webrtc {
// RemoteBitrateObserver is used to signal changes in bitrate estimates for
// the incoming stream.
class RemoteBitrateObserver {
public:
// Called when a receive channel has a new bitrate estimate for the incoming
// stream.
virtual void OnReceiveBitrateChanged(unsigned int ssrc,
unsigned int bitrate) = 0;
virtual ~RemoteBitrateObserver() {}
};
class RemoteBitrateEstimator {
public:
explicit RemoteBitrateEstimator(RemoteBitrateObserver* observer);
// Called for each incoming packet. If this is a new SSRC, a new
// BitrateControl will be created.
void IncomingPacket(unsigned int ssrc,
int packet_size,
int64_t arrival_time,
uint32_t rtp_timestamp,
int64_t packet_send_time);
// Triggers a new estimate calculation for the stream identified by |ssrc|.
void UpdateEstimate(unsigned int ssrc, int64_t time_now);
// Set the current round-trip time experienced by the stream identified by
// |ssrc|.
void SetRtt(unsigned int ssrc);
// Removes all data for |ssrc|.
void RemoveStream(unsigned int ssrc);
// Returns true if a valid estimate exists for a stream identified by |ssrc|
// and sets |bitrate_bps| to the estimated bitrate in bits per second.
bool LatestEstimate(unsigned int ssrc, unsigned int* bitrate_bps) const;
private:
struct BitrateControls {
RemoteRateControl remote_rate;
OverUseDetector overuse_detector;
BitRateStats incoming_bitrate;
};
typedef std::map<unsigned int, BitrateControls> SsrcBitrateControlsMap;
SsrcBitrateControlsMap bitrate_controls_;
RemoteBitrateObserver* observer_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_

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src/modules/remote_bitrate_estimator/overuse_detector.cc
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/*
* 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 <math.h>
#include <stdlib.h> // abs
#if _WIN32
#include <windows.h>
#endif
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "system_wrappers/interface/trace.h"
#ifdef WEBRTC_BWE_MATLAB
extern MatlabEngine eng; // global variable defined elsewhere
#endif
#define INIT_CAPACITY_SLOPE 8.0/512.0
#define DETECTOR_THRESHOLD 25.0
#define OVER_USING_TIME_THRESHOLD 100
#define MIN_FRAME_PERIOD_HISTORY_LEN 60
namespace webrtc {
OverUseDetector::OverUseDetector()
: firstPacket_(true),
currentFrame_(),
prevFrame_(),
numOfDeltas_(0),
slope_(INIT_CAPACITY_SLOPE),
offset_(0),
E_(),
processNoise_(),
avgNoise_(0.0),
varNoise_(500),
threshold_(DETECTOR_THRESHOLD),
tsDeltaHist_(),
prevOffset_(0.0),
timeOverUsing_(-1),
overUseCounter_(0),
#ifndef WEBRTC_BWE_MATLAB
hypothesis_(kBwNormal) {
#else
plot1_(NULL),
plot2_(NULL),
plot3_(NULL),
plot4_(NULL) {
#endif
E_[0][0] = 100;
E_[1][1] = 1e-1;
E_[0][1] = E_[1][0] = 0;
processNoise_[0] = 1e-10;
processNoise_[1] = 1e-2;
}
OverUseDetector::~OverUseDetector() {
#ifdef WEBRTC_BWE_MATLAB
if (plot1_) {
eng.DeletePlot(plot1_);
plot1_ = NULL;
}
if (plot2_) {
eng.DeletePlot(plot2_);
plot2_ = NULL;
}
if (plot3_) {
eng.DeletePlot(plot3_);
plot3_ = NULL;
}
if (plot4_) {
eng.DeletePlot(plot4_);
plot4_ = NULL;
}
#endif
tsDeltaHist_.clear();
}
void OverUseDetector::Reset() {
firstPacket_ = true;
currentFrame_.size_ = 0;
currentFrame_.completeTimeMs_ = -1;
currentFrame_.timestamp_ = -1;
prevFrame_.size_ = 0;
prevFrame_.completeTimeMs_ = -1;
prevFrame_.timestamp_ = -1;
numOfDeltas_ = 0;
slope_ = INIT_CAPACITY_SLOPE;
offset_ = 0;
E_[0][0] = 100;
E_[1][1] = 1e-1;
E_[0][1] = E_[1][0] = 0;
processNoise_[0] = 1e-10;
processNoise_[1] = 1e-2;
avgNoise_ = 0.0;
varNoise_ = 500;
threshold_ = DETECTOR_THRESHOLD;
prevOffset_ = 0.0;
timeOverUsing_ = -1;
overUseCounter_ = 0;
hypothesis_ = kBwNormal;
tsDeltaHist_.clear();
}
void OverUseDetector::Update(WebRtc_UWord16 packetSize,
WebRtc_UWord32 timestamp,
const WebRtc_Word64 nowMS) {
#ifdef WEBRTC_BWE_MATLAB
// Create plots
const WebRtc_Word64 startTimeMs = nowMS;
if (plot1_ == NULL) {
plot1_ = eng.NewPlot(new MatlabPlot());
plot1_->AddLine(1000, "b.", "scatter");
}
if (plot2_ == NULL) {
plot2_ = eng.NewPlot(new MatlabPlot());
plot2_->AddTimeLine(30, "b", "offset", startTimeMs);
plot2_->AddTimeLine(30, "r--", "limitPos", startTimeMs);
plot2_->AddTimeLine(30, "k.", "trigger", startTimeMs);
plot2_->AddTimeLine(30, "ko", "detection", startTimeMs);
// plot2_->AddTimeLine(30, "g", "slowMean", startTimeMs);
}
if (plot3_ == NULL) {
plot3_ = eng.NewPlot(new MatlabPlot());
plot3_->AddTimeLine(30, "b", "noiseVar", startTimeMs);
}
if (plot4_ == NULL) {
plot4_ = eng.NewPlot(new MatlabPlot());
// plot4_->AddTimeLine(60, "b", "p11", startTimeMs);
// plot4_->AddTimeLine(60, "r", "p12", startTimeMs);
plot4_->AddTimeLine(60, "g", "p22", startTimeMs);
// plot4_->AddTimeLine(60, "g--", "p22_hat", startTimeMs);
// plot4_->AddTimeLine(30, "b.-", "deltaFs", startTimeMs);
}
#endif
bool wrapped = false;
bool completeFrame = false;
if (currentFrame_.timestamp_ == -1) {
currentFrame_.timestamp_ = timestamp;
} else if (OldTimestamp(
timestamp,
static_cast<WebRtc_UWord32>(currentFrame_.timestamp_),
&wrapped)) {
// Don't update with old data
return;
} 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

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/*
* 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_

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/*
* 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

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src/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
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# 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:

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src/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest.cc
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/*
* 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

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/*
* 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

83
src/modules/remote_bitrate_estimator/remote_rate_control.h
View File

@ -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
View 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:

7
src/modules/remote_bitrate_estimator/bitrate_estimator_unittest.cc → src/modules/rtp_rtcp/test/test_bwe/unit_test.cc
View File

@ -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 {

2
tools/continuous_build/build_internal/masters/master.webrtc/master.cfg
View File

@ -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),