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Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module.

Browse Source 
Each REMB group has one remote_bitrate_estimator object. For now the
estimator keeps one estimate for every SSRC. In a later commit this will
be unified and one estimate will be used for all SSRC in one group.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/605007

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/634006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2366 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2012-06-05 12:25:35 +00:00
parent df373984e2
commit cb89c6f914
31 changed files with 1455 additions and 136 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
src/modules/modules.gyp
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@ -28,7 +28,6 @@
'audio_processing/utility/util.gypi',
'bitrate_controller/bitrate_controller.gypi',
'media_file/source/media_file.gypi',
'remote_bitrate_estimator/remote_bitrate_estimator.gypi',
'udp_transport/source/udp_transport.gypi',
'utility/source/utility.gypi',
'video_coding/codecs/i420/main/source/i420.gypi',
@ -48,6 +47,7 @@
'audio_coding/codecs/iSAC/isacfix_test.gypi',
'audio_processing/apm_tests.gypi',
'rtp_rtcp/source/rtp_rtcp_tests.gypi',
'rtp_rtcp/test/test_bwe/test_bwe.gypi',
'rtp_rtcp/test/testFec/test_fec.gypi',
'rtp_rtcp/test/testAPI/test_api.gypi',
'video_coding/main/source/video_coding_test.gypi',

10
src/modules/rtp_rtcp/interface/rtp_rtcp.h
View File

@ -18,8 +18,6 @@
namespace webrtc {
// forward declaration
class RemoteBitrateEstimator;
class RemoteBitrateObserver;
class Transport;
class RtpRtcp : public Module {
@ -37,7 +35,7 @@ class RtpRtcp : public Module {
intra_frame_callback(NULL),
bandwidth_callback(NULL),
audio_messages(NULL),
remote_bitrate_estimator(NULL) {
bitrate_observer(NULL) {
}
/* id - Unique identifier of this RTP/RTCP module object
* audio - True for a audio version of the RTP/RTCP module
@ -56,8 +54,8 @@ class RtpRtcp : public Module {
* bandwidth_callback - Called when we receive a changed estimate from
* the receiver of out stream.
* audio_messages - Telehone events.
* remote_bitrate_estimator - Estimates the bandwidth available for a set of
* streams from the same client.
* bitrate_observer - Called when the estimate of the incoming RTP
* stream changes.
*/
int32_t id;
bool audio;
@ -70,7 +68,7 @@ class RtpRtcp : public Module {
RtcpIntraFrameObserver* intra_frame_callback;
RtcpBandwidthObserver* bandwidth_callback;
RtpAudioFeedback* audio_messages;
RemoteBitrateEstimator* remote_bitrate_estimator;
RtpRemoteBitrateObserver* bitrate_observer;
};
/*
* Create a RTP/RTCP module object using the system clock.

11
src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
View File

@ -251,5 +251,16 @@ class RtpRtcpClock {
virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) = 0;
};
// RtpReceiveBitrateUpdate is used to signal changes in bitrate estimates for
// the incoming stream.
class RtpRemoteBitrateObserver {
public:
// Called when a receive channel has a new bitrate estimate for the incoming
// stream.
virtual void OnReceiveBitrateChanged(uint32_t ssrc,
uint32_t bitrate) = 0;
virtual ~RtpRemoteBitrateObserver() {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_

9
src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
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@ -8,10 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
#include <gmock/gmock.h>
#include "../testing/gmock/include/gmock/gmock.h"
#include "modules/interface/module.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
@ -214,7 +211,7 @@ class MockRtpRtcp : public RtpRtcp {
MOCK_METHOD3(SetREMBData,
WebRtc_Word32(const WebRtc_UWord32 bitrate, const WebRtc_UWord8 numberOfSSRC, const WebRtc_UWord32* SSRC));
MOCK_METHOD1(SetRemoteBitrateObserver,
bool(RemoteBitrateObserver*));
bool(RtpRemoteBitrateObserver*));
MOCK_CONST_METHOD0(IJ,
bool());
MOCK_METHOD1(SetIJStatus,
@ -288,5 +285,3 @@ class MockRtpRtcp : public RtpRtcp {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_

28
src/modules/rtp_rtcp/source/Bitrate.h
View File

@ -55,6 +55,32 @@ private:
WebRtc_UWord32 _packetCount;
};
} // namespace webrtc
struct DataTimeSizeTuple
{
DataTimeSizeTuple(WebRtc_UWord32 sizeBytes, WebRtc_Word64 timeCompleteMs) :
_sizeBytes(sizeBytes),
_timeCompleteMs(timeCompleteMs) {}
WebRtc_UWord32 _sizeBytes;
WebRtc_Word64 _timeCompleteMs;
};
class BitRateStats
{
public:
BitRateStats();
~BitRateStats();
void Init();
void Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs);
WebRtc_UWord32 BitRate(WebRtc_Word64 nowMs);
private:
void EraseOld(WebRtc_Word64 nowMs);
std::list<DataTimeSizeTuple*> _dataSamples;
WebRtc_UWord32 _accumulatedBytes;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_

75
src/modules/rtp_rtcp/source/bitrate.cc
View File

@ -11,6 +11,8 @@
#include "Bitrate.h"
#include "rtp_utility.h"
#define BITRATE_AVERAGE_WINDOW 2000
namespace webrtc {
Bitrate::Bitrate(RtpRtcpClock* clock) :
_clock(*clock),
@ -110,4 +112,77 @@ Bitrate::Process()
}
}
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 >
BITRATE_AVERAGE_WINDOW)
{
// 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

56
src/modules/rtp_rtcp/source/bwe_defines.h Normal file
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@ -0,0 +1,56 @@
/*
* 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_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_

6
src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
View File

@ -8,16 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_
#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
namespace webrtc {
class MockRTPReceiverVideo : public RTPReceiverVideo {
public:
MockRTPReceiverVideo() : RTPReceiverVideo(0, NULL, NULL) {}
MOCK_METHOD1(ChangeUniqueId,
void(const WebRtc_Word32 id));
MOCK_METHOD3(ReceiveRecoveredPacketCallback,
@ -43,5 +39,3 @@ class MockRTPReceiverVideo : public RTPReceiverVideo {
};
} // namespace webrtc
#endif //WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_RECEIVER_VIDEO_H_

423
src/modules/rtp_rtcp/source/overuse_detector.cc Normal file
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@ -0,0 +1,423 @@
/*
* 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/rtp_rtcp/source/overuse_detector.h"
#include "modules/rtp_rtcp/source/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();
}
bool OverUseDetector::Update(const WebRtcRTPHeader& rtpHeader,
const WebRtc_UWord16 packetSize,
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_ = rtpHeader.header.timestamp;
} else if (ModuleRTPUtility::OldTimestamp(
rtpHeader.header.timestamp,
static_cast<WebRtc_UWord32>(currentFrame_.timestamp_),
&wrapped)) {
// Don't update with old data
return completeFrame;
} else if (rtpHeader.header.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
ModuleRTPUtility::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_ = rtpHeader.header.timestamp;
currentFrame_.size_ = 0;
currentFrame_.completeTimeMs_ = -1;
completeFrame = true;
}
// Accumulate the frame size
currentFrame_.size_ += packetSize;
currentFrame_.completeTimeMs_ = nowMS;
return completeFrame;
}
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_;
}
} // namespace webrtc

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src/modules/rtp_rtcp/source/overuse_detector.h Normal file
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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/rtp_rtcp/source/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();
bool Update(const WebRtcRTPHeader& rtpHeader,
const WebRtc_UWord16 packetSize,
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_;
};
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_

483
src/modules/rtp_rtcp/source/remote_rate_control.cc Normal file
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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.
*/
#if _WIN32
#include <windows.h>
#endif
#include "remote_rate_control.h"
#include "trace.h"
#include <math.h>
#include <string.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)
#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_UWord32 RTT,
WebRtc_Word64 nowMS)
{
_currentBitRate = ChangeBitRate(_currentBitRate, _currentInput._incomingBitRate,
_currentInput._noiseVar, RTT, nowMS);
return _currentBitRate;
}
RateControlRegion RemoteRateControl::Update(const RateControlInput& input,
bool& firstOverUse,
WebRtc_Word64 nowMS)
{
#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
firstOverUse = (_currentInput._bwState != kBwOverusing &&
input._bwState == kBwOverusing);
// 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_UWord32 RTT,
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

82
src/modules/rtp_rtcp/source/remote_rate_control.h Normal file
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@ -0,0 +1,82 @@
/*
* 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 "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_UWord32 RTT,
WebRtc_Word64 nowMS);
RateControlRegion Update(const RateControlInput& input, bool& firstOverUse,
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_UWord32 RTT,
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;
#ifdef MATLAB
MatlabPlot *_plot1;
MatlabPlot *_plot2;
#endif
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_

10
src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
View File

@ -16,8 +16,7 @@
#include "rtcp_sender.h"
#include "rtcp_receiver.h"
#include "rtp_rtcp_impl.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "bwe_defines.h"
namespace {
@ -58,9 +57,7 @@ class TestTransport : public Transport {
class RtcpFormatRembTest : public ::testing::Test {
protected:
RtcpFormatRembTest()
: remote_bitrate_observer_(),
remote_bitrate_estimator_(&remote_bitrate_observer_) {}
RtcpFormatRembTest() {};
virtual void SetUp();
virtual void TearDown();
@ -69,8 +66,6 @@ class RtcpFormatRembTest : public ::testing::Test {
RTCPSender* rtcp_sender_;
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
MockRemoteBitrateObserver remote_bitrate_observer_;
RemoteBitrateEstimator remote_bitrate_estimator_;
};
void RtcpFormatRembTest::SetUp() {
@ -79,7 +74,6 @@ void RtcpFormatRembTest::SetUp() {
configuration.id = 0;
configuration.audio = false;
configuration.clock = system_clock_;
configuration.remote_bitrate_estimator = &remote_bitrate_estimator_;
dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
rtcp_sender_ = new RTCPSender(0, false, system_clock_, dummy_rtp_rtcp_impl_);
rtcp_receiver_ = new RTCPReceiver(0, system_clock_, dummy_rtp_rtcp_impl_);

11
src/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
View File

@ -12,13 +12,10 @@
/*
* This file includes unit tests for the RTCPReceiver.
*/
#include <gmock/gmock.h>
#include <gtest/gtest.h>
// Note: This file has no directory. Lint warning must be ignored.
#include "common_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
@ -182,19 +179,15 @@ class TestTransport : public Transport,
class RtcpReceiverTest : public ::testing::Test {
protected:
RtcpReceiverTest()
: remote_bitrate_observer_(),
remote_bitrate_estimator_(&remote_bitrate_observer_) {
RtcpReceiverTest() {
// system_clock_ = ModuleRTPUtility::GetSystemClock();
system_clock_ = new FakeSystemClock();
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
configuration.id = 0;
configuration.audio = false;
configuration.clock = system_clock_;
configuration.outgoing_transport = test_transport_;
configuration.remote_bitrate_estimator = &remote_bitrate_estimator_;
rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
test_transport_->SetRTCPReceiver(rtcp_receiver_);
@ -226,8 +219,6 @@ class RtcpReceiverTest : public ::testing::Test {
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
MockRemoteBitrateObserver remote_bitrate_observer_;
RemoteBitrateEstimator remote_bitrate_estimator_;
};

62
src/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -10,15 +10,15 @@
#include "rtcp_sender.h"
#include <cassert> // assert
#include <cstdlib> // rand
#include <string.h> // memcpy
#include <string.h> // memcpy
#include <cassert> // assert
#include <cstdlib> // rand
#include "trace.h"
#include "common_types.h"
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "system_wrappers/interface/critical_section_wrapper.h"
#include "system_wrappers/interface/trace.h"
#include "critical_section_wrapper.h"
#include "rtp_rtcp_impl.h"
namespace webrtc {
@ -66,10 +66,12 @@ RTCPSender::RTCPSender(const WebRtc_Word32 id,
_sizeRembSSRC(0),
_rembSSRC(NULL),
_rembBitrate(0),
_bitrate_observer(NULL),
_tmmbrHelp(),
_tmmbr_Send(0),
_packetOH_Send(0),
_remoteRateControl(),
_appSend(false),
_appSubType(0),
@ -128,7 +130,7 @@ RTCPSender::Init()
_sequenceNumberFIR = 0;
_tmmbr_Send = 0;
_packetOH_Send = 0;
//_remoteRateControl.Reset();
_remoteRateControl.Reset();
_nextTimeToSendRTCP = 0;
_CSRCs = 0;
_appSend = false;
@ -259,6 +261,22 @@ RTCPSender::SetREMBData(const WebRtc_UWord32 bitrate,
return 0;
}
bool RTCPSender::SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (observer && _bitrate_observer) {
return false;
}
_bitrate_observer = observer;
return true;
}
void RTCPSender::UpdateRemoteBitrateEstimate(unsigned int target_bitrate) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (_bitrate_observer) {
_bitrate_observer->OnReceiveBitrateChanged(_remoteSSRC, target_bitrate);
}
}
bool
RTCPSender::TMMBR() const
{
@ -309,7 +327,7 @@ RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_remoteSSRC = ssrc;
//_remoteRateControl.Reset();
_remoteRateControl.Reset();
return 0;
}
@ -1086,11 +1104,25 @@ RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
return 0;
}
void
RTCPSender::SetTargetBitrate(unsigned int target_bitrate)
WebRtc_UWord32
RTCPSender::CalculateNewTargetBitrate(WebRtc_UWord32 RTT)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
WebRtc_UWord32 target_bitrate =
_remoteRateControl.UpdateBandwidthEstimate(RTT, _clock.GetTimeInMS());
_tmmbr_Send = target_bitrate / 1000;
return target_bitrate;
}
WebRtc_UWord32 RTCPSender::LatestBandwidthEstimate() const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _remoteRateControl.LatestEstimate();
}
bool
RTCPSender::ValidBitrateEstimate() const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _remoteRateControl.ValidEstimate();
}
WebRtc_Word32
@ -2131,4 +2163,12 @@ RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
}
return -1;
}
RateControlRegion
RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _remoteRateControl.Update(rateControlInput, firstOverUse,
_clock.GetTimeInMS());
}
} // namespace webrtc

24
src/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -17,10 +17,8 @@
#include "rtcp_utility.h"
#include "rtp_utility.h"
#include "rtp_rtcp_defines.h"
#include "scoped_ptr.h"
#include "remote_rate_control.h"
#include "tmmbr_help.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
namespace webrtc {
@ -89,6 +87,10 @@ public:
const WebRtc_UWord8 numberOfSSRC,
const WebRtc_UWord32* SSRC);
bool SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer);
void UpdateRemoteBitrateEstimate(unsigned int target_bitrate);
/*
* TMMBR
*/
@ -122,7 +124,19 @@ public:
WebRtc_Word32 SetCSRCStatus(const bool include);
void SetTargetBitrate(unsigned int target_bitrate);
/*
* New bandwidth estimation
*/
RateControlRegion UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse);
WebRtc_UWord32 CalculateNewTargetBitrate(WebRtc_UWord32 RTT);
WebRtc_UWord32 LatestBandwidthEstimate() const;
// Returns true if there is a valid estimate of the incoming bitrate, false
// otherwise.
bool ValidBitrateEstimate() const;
private:
WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
@ -226,10 +240,12 @@ private:
WebRtc_UWord8 _sizeRembSSRC;
WebRtc_UWord32* _rembSSRC;
WebRtc_UWord32 _rembBitrate;
RtpRemoteBitrateObserver* _bitrate_observer;
TMMBRHelp _tmmbrHelp;
WebRtc_UWord32 _tmmbr_Send;
WebRtc_UWord32 _packetOH_Send;
RemoteRateControl _remoteRateControl;
// APP
bool _appSend;

18
src/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -13,16 +13,13 @@
* This file includes unit tests for the RTCPSender.
*/
#include <gmock/gmock.h>
#include <gtest/gtest.h>
#include "common_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "rtp_utility.h"
#include "rtcp_sender.h"
#include "rtcp_receiver.h"
#include "rtp_rtcp_impl.h"
namespace webrtc {
@ -97,9 +94,7 @@ class TestTransport : public Transport,
class RtcpSenderTest : public ::testing::Test {
protected:
RtcpSenderTest()
: remote_bitrate_observer_(),
remote_bitrate_estimator_(&remote_bitrate_observer_) {
RtcpSenderTest() {
system_clock_ = ModuleRTPUtility::GetSystemClock();
test_transport_ = new TestTransport();
@ -109,7 +104,6 @@ class RtcpSenderTest : public ::testing::Test {
configuration.clock = system_clock_;
configuration.incoming_data = test_transport_;
configuration.outgoing_transport = test_transport_;
configuration.remote_bitrate_estimator = &remote_bitrate_estimator_;
rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
rtcp_sender_ = new RTCPSender(0, false, system_clock_, rtp_rtcp_impl_);
@ -138,8 +132,6 @@ class RtcpSenderTest : public ::testing::Test {
RTCPSender* rtcp_sender_;
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
MockRemoteBitrateObserver remote_bitrate_observer_;
RemoteBitrateEstimator remote_bitrate_estimator_;
enum {kMaxPacketLength = 1500};
uint8_t packet_[kMaxPacketLength];

4
src/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@ -32,10 +32,9 @@ using ModuleRTPUtility::VideoPayload;
RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
const bool audio,
RtpRtcpClock* clock,
RemoteBitrateEstimator* remote_bitrate,
ModuleRtpRtcpImpl* owner) :
RTPReceiverAudio(id),
RTPReceiverVideo(id, remote_bitrate, owner),
RTPReceiverVideo(id, owner),
Bitrate(clock),
_id(id),
_audio(audio),
@ -1084,6 +1083,7 @@ void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader) {
// reset last report
ResetStatistics();
RTPReceiverVideo::ResetOverUseDetector();
_lastReceivedTimestamp = 0;
_lastReceivedSequenceNumber = 0;

1
src/modules/rtp_rtcp/source/rtp_receiver.h
View File

@ -35,7 +35,6 @@ public:
RTPReceiver(const WebRtc_Word32 id,
const bool audio,
RtpRtcpClock* clock,
RemoteBitrateEstimator* remote_bitrate,
ModuleRtpRtcpImpl* owner);
virtual ~RTPReceiver();

51
src/modules/rtp_rtcp/source/rtp_receiver_video.cc
View File

@ -26,8 +26,20 @@ WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
}
RTPReceiverVideo::RTPReceiverVideo()
: _id(0),
_rtpRtcp(NULL),
_criticalSectionReceiverVideo(
CriticalSectionWrapper::CreateCriticalSection()),
_currentFecFrameDecoded(false),
_receiveFEC(NULL),
_overUseDetector(),
_videoBitRate(),
_lastBitRateChange(0),
_packetOverHead(28) {
}
RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
RemoteBitrateEstimator* remote_bitrate,
ModuleRtpRtcpImpl* owner)
: _id(id),
_rtpRtcp(owner),
@ -35,7 +47,9 @@ RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
CriticalSectionWrapper::CreateCriticalSection()),
_currentFecFrameDecoded(false),
_receiveFEC(NULL),
remote_bitrate_(remote_bitrate),
_overUseDetector(),
_videoBitRate(),
_lastBitRateChange(0),
_packetOverHead(28) {
}
@ -73,6 +87,12 @@ ModuleRTPUtility::Payload* RTPReceiverVideo::RegisterReceiveVideoPayload(
return payload;
}
void RTPReceiverVideo::ResetOverUseDetector() {
_overUseDetector.Reset();
_videoBitRate.Init();
_lastBitRateChange = 0;
}
// we have no critext when calling this
// we are not allowed to have any critsects when calling
// CallbackOfReceivedPayloadData
@ -89,15 +109,14 @@ WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecific(
_criticalSectionReceiverVideo->Enter();
_videoBitRate.Update(payloadDataLength + rtpHeader->header.paddingLength,
nowMS);
// Add headers, ideally we would like to include for instance
// Ethernet header here as well.
const WebRtc_UWord16 packetSize = payloadDataLength + _packetOverHead +
rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
remote_bitrate_->IncomingPacket(rtpHeader->header.ssrc,
packetSize,
nowMS,
rtpHeader->header.timestamp,
-1);
_overUseDetector.Update(*rtpHeader, packetSize, nowMS);
if (isRED) {
if(_receiveFEC == NULL) {
@ -135,6 +154,24 @@ WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecific(
payloadDataLength,
videoType);
}
// Update the remote rate control object and update the overuse
// detector with the current rate control region.
_criticalSectionReceiverVideo->Enter();
const RateControlInput input(_overUseDetector.State(),
_videoBitRate.BitRate(nowMS),
_overUseDetector.NoiseVar());
_criticalSectionReceiverVideo->Leave();
// Call the callback outside critical section
if (_rtpRtcp) {
const RateControlRegion region = _rtpRtcp->OnOverUseStateUpdate(input);
_criticalSectionReceiverVideo->Enter();
_overUseDetector.SetRateControlRegion(region);
_criticalSectionReceiverVideo->Leave();
}
return retVal;
}

17
src/modules/rtp_rtcp/source/rtp_receiver_video.h
View File

@ -16,11 +16,9 @@
#include "typedefs.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/remote_rate_control.h"
#include "overuse_detector.h"
#include "remote_rate_control.h"
#include "Bitrate.h"
#include "scoped_ptr.h"
namespace webrtc {
class ReceiverFEC;
@ -29,9 +27,8 @@ class CriticalSectionWrapper;
class RTPReceiverVideo {
public:
RTPReceiverVideo(const WebRtc_Word32 id,
RemoteBitrateEstimator* remote_bitrate,
ModuleRtpRtcpImpl* owner);
RTPReceiverVideo();
RTPReceiverVideo(const WebRtc_Word32 id, ModuleRtpRtcpImpl* owner);
virtual ~RTPReceiverVideo();
@ -58,6 +55,8 @@ class RTPReceiverVideo {
void SetPacketOverHead(WebRtc_UWord16 packetOverHead);
protected:
void ResetOverUseDetector();
virtual WebRtc_Word32 CallbackOfReceivedPayloadData(
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
@ -107,7 +106,9 @@ class RTPReceiverVideo {
ReceiverFEC* _receiveFEC;
// BWE
RemoteBitrateEstimator* remote_bitrate_;
OverUseDetector _overUseDetector;
BitRateStats _videoBitRate;
WebRtc_Word64 _lastBitRateChange;
WebRtc_UWord16 _packetOverHead;
};
} // namespace webrtc

5
src/modules/rtp_rtcp/source/rtp_rtcp.gypi
View File

@ -13,7 +13,6 @@
'type': '<(library)',
'dependencies': [
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
],
'include_dirs': [
'../interface',
@ -68,8 +67,12 @@
'forward_error_correction.h',
'forward_error_correction_internal.cc',
'forward_error_correction_internal.h',
'overuse_detector.cc',
'overuse_detector.h',
'producer_fec.cc',
'producer_fec.h',
'remote_rate_control.cc',
'remote_rate_control.h',
'rtp_packet_history.cc',
'rtp_packet_history.h',
'rtp_receiver_video.cc',

49
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -54,7 +54,7 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: _rtpSender(configuration.id, configuration.audio, configuration.clock),
_rtpReceiver(configuration.id, configuration.audio, configuration.clock,
configuration.remote_bitrate_estimator, this),
this),
_rtcpSender(configuration.id, configuration.audio, configuration.clock,
this),
_rtcpReceiver(configuration.id, configuration.clock, this),
@ -80,8 +80,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
_nackLastTimeSent(0),
_nackLastSeqNumberSent(0),
_simulcast(false),
_keyFrameReqMethod(kKeyFrameReqFirRtp),
remote_bitrate_(configuration.remote_bitrate_estimator)
_keyFrameReqMethod(kKeyFrameReqFirRtp)
#ifdef MATLAB
, _plot1(NULL)
#endif
@ -103,6 +102,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
_rtpSender.RegisterSendTransport(configuration.outgoing_transport);
_rtcpSender.RegisterSendTransport(configuration.outgoing_transport);
_rtcpSender.SetRemoteBitrateObserver(configuration.bitrate_observer);
// make sure that RTCP objects are aware of our SSRC
WebRtc_UWord32 SSRC = _rtpSender.SSRC();
_rtcpSender.SetSSRC(SSRC);
@ -223,16 +224,15 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
// default module or no RTCP received yet.
max_rtt = kDefaultRtt;
}
remote_bitrate_->SetRtt(max_rtt);
remote_bitrate_->UpdateEstimate(_rtpReceiver.SSRC(), now);
if (TMMBR()) {
unsigned int target_bitrate = 0;
if (remote_bitrate_->LatestEstimate(_rtpReceiver.SSRC(),
&target_bitrate)) {
_rtcpSender.SetTargetBitrate(target_bitrate);
if (_rtcpSender.ValidBitrateEstimate()) {
if (REMB()) {
uint32_t target_bitrate =
_rtcpSender.CalculateNewTargetBitrate(max_rtt);
_rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
} else if (TMMBR()) {
_rtcpSender.CalculateNewTargetBitrate(max_rtt);
}
}
_rtcpSender.SendRTCP(kRtcpReport);
}
@ -1882,12 +1882,35 @@ void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* totalRate,
int ModuleRtpRtcpImpl::EstimatedReceiveBandwidth(
WebRtc_UWord32* available_bandwidth) const {
if (!remote_bitrate_->LatestEstimate(_rtpReceiver.SSRC(),
available_bandwidth))
if (!_rtcpSender.ValidBitrateEstimate())
return -1;
*available_bandwidth = _rtcpSender.LatestBandwidthEstimate();
return 0;
}
RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
const RateControlInput& rateControlInput) {
bool firstOverUse = false;
RateControlRegion region = _rtcpSender.UpdateOverUseState(rateControlInput,
firstOverUse);
if (firstOverUse) {
// Send TMMBR or REMB immediately.
WebRtc_UWord16 RTT = 0;
_rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
// About to send TMMBR, first run remote rate control
// to get a target bit rate.
unsigned int target_bitrate =
_rtcpSender.CalculateNewTargetBitrate(RTT);
if (REMB()) {
_rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
} else if (TMMBR()) {
_rtcpSender.SendRTCP(kRtcpTmmbr);
}
}
return region;
}
// bad state of RTP receiver request a keyframe
void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
RequestKeyFrame();

4
src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -428,6 +428,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
virtual RateControlRegion OnOverUseStateUpdate(const RateControlInput& rateControlInput);
// good state of RTP receiver inform sender
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
@ -504,8 +506,6 @@ private:
VideoCodec _sendVideoCodec;
KeyFrameRequestMethod _keyFrameReqMethod;
RemoteBitrateEstimator* remote_bitrate_;
#ifdef MATLAB
MatlabPlot* _plot1;
#endif

4
src/video_engine/vie_channel.cc
View File

@ -38,7 +38,7 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
ProcessThread& module_process_thread,
RtcpIntraFrameObserver* intra_frame_observer,
RtcpBandwidthObserver* bandwidth_observer,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpRemoteBitrateObserver* bitrate_observer,
RtpRtcp* default_rtp_rtcp)
: ViEFrameProviderBase(channel_id, engine_id),
channel_id_(channel_id),
@ -91,7 +91,7 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
configuration.rtcp_feedback = this;
configuration.intra_frame_callback = intra_frame_observer;
configuration.bandwidth_callback = bandwidth_observer;
configuration.remote_bitrate_estimator = remote_bitrate_estimator;
configuration.bitrate_observer = bitrate_observer;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());

3
src/video_engine/vie_channel.h
View File

@ -15,7 +15,6 @@
#include <list>
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "modules/udp_transport/interface/udp_transport.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
@ -64,7 +63,7 @@ class ViEChannel
ProcessThread& module_process_thread,
RtcpIntraFrameObserver* intra_frame_observer,
RtcpBandwidthObserver* bandwidth_observer,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpRemoteBitrateObserver* bitrate_observer,
RtpRtcp* default_rtp_rtcp);
~ViEChannel();

17
src/video_engine/vie_channel_group.cc
View File

@ -11,7 +11,6 @@
#include "video_engine/vie_channel_group.h"
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "video_engine/vie_channel.h"
#include "video_engine/vie_encoder.h"
@ -21,8 +20,7 @@ namespace webrtc {
ChannelGroup::ChannelGroup(ProcessThread* process_thread)
: remb_(new VieRemb(process_thread)),
bitrate_controller_(BitrateController::CreateBitrateController()),
remote_bitrate_estimator_(new RemoteBitrateEstimator(remb_.get())) {
bitrate_controller_(BitrateController::CreateBitrateController()) {
}
ChannelGroup::~ChannelGroup() {
@ -33,9 +31,8 @@ void ChannelGroup::AddChannel(int channel_id) {
channels_.insert(channel_id);
}
void ChannelGroup::RemoveChannel(int channel_id, unsigned int ssrc) {
void ChannelGroup::RemoveChannel(int channel_id) {
channels_.erase(channel_id);
remote_bitrate_estimator_->RemoveStream(ssrc);
}
bool ChannelGroup::HasChannel(int channel_id) {
@ -46,12 +43,12 @@ bool ChannelGroup::Empty() {
return channels_.empty();
}
BitrateController* ChannelGroup::GetBitrateController() {
return bitrate_controller_.get();
RtpRemoteBitrateObserver* ChannelGroup::GetRtpRemoteBitrateObserver() {
return remb_.get();
}
RemoteBitrateEstimator* ChannelGroup::GetRemoteBitrateEstimator() {
return remote_bitrate_estimator_.get();
BitrateController* ChannelGroup::GetBitrateController() {
return bitrate_controller_.get();
}
bool ChannelGroup::SetChannelRembStatus(int channel_id,
@ -67,7 +64,7 @@ bool ChannelGroup::SetChannelRembStatus(int channel_id,
} else if (channel) {
channel->EnableRemb(false);
}
// Update the REMB instance with necessary RTP modules.
// Update the remb instance with necesary RTp modules.
RtpRtcp* rtp_module = channel->rtp_rtcp();
if (sender) {
remb_->AddRembSender(rtp_module);

9
src/video_engine/vie_channel_group.h
View File

@ -19,8 +19,7 @@ namespace webrtc {
class BitrateController;
class ProcessThread;
class RemoteBitrateEstimator;
class RemoteBitrateObserver;
class RtpRemoteBitrateObserver;
class ViEChannel;
class ViEEncoder;
class VieRemb;
@ -33,7 +32,7 @@ class ChannelGroup {
~ChannelGroup();
void AddChannel(int channel_id);
void RemoveChannel(int channel_id, unsigned int ssrc);
void RemoveChannel(int channel_id);
bool HasChannel(int channel_id);
bool Empty();
@ -44,14 +43,14 @@ class ChannelGroup {
ViEEncoder* encoder);
BitrateController* GetBitrateController();
RemoteBitrateEstimator* GetRemoteBitrateEstimator();
RtpRemoteBitrateObserver* GetRtpRemoteBitrateObserver();
private:
typedef std::set<int> ChannelSet;
scoped_ptr<VieRemb> remb_;
scoped_ptr<BitrateController> bitrate_controller_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
ChannelSet channels_;
};

25
src/video_engine/vie_channel_manager.cc
View File

@ -97,12 +97,13 @@ int ViEChannelManager::CreateChannel(int& channel_id) {
RtcpBandwidthObserver* bandwidth_observer =
bitrate_controller->CreateRtcpBandwidthObserver();
RemoteBitrateEstimator* remote_bitrate_estimator =
group->GetRemoteBitrateEstimator();
RtpRemoteBitrateObserver* bitrate_observer =
group->GetRtpRemoteBitrateObserver();
if (!(vie_encoder->Init() &&
CreateChannelObject(new_channel_id, vie_encoder, bandwidth_observer,
remote_bitrate_estimator))) {
bitrate_observer))) {
delete vie_encoder;
vie_encoder = NULL;
ReturnChannelId(new_channel_id);
@ -135,8 +136,9 @@ int ViEChannelManager::CreateChannel(int& channel_id,
RtcpBandwidthObserver* bandwidth_observer =
bitrate_controller->CreateRtcpBandwidthObserver();
RemoteBitrateEstimator* remote_bitrate_estimator =
channel_group->GetRemoteBitrateEstimator();
RtpRemoteBitrateObserver* bitrate_observer =
channel_group->GetRtpRemoteBitrateObserver();
ViEEncoder* vie_encoder = NULL;
if (sender) {
@ -146,8 +148,7 @@ int ViEChannelManager::CreateChannel(int& channel_id,
bitrate_controller);
if (!(vie_encoder->Init() &&
CreateChannelObject(new_channel_id, vie_encoder,
bandwidth_observer,
remote_bitrate_estimator))) {
bandwidth_observer, bitrate_observer))) {
delete vie_encoder;
vie_encoder = NULL;
}
@ -155,7 +156,7 @@ int ViEChannelManager::CreateChannel(int& channel_id,
vie_encoder = ViEEncoderPtr(original_channel);
assert(vie_encoder);
if (!CreateChannelObject(new_channel_id, vie_encoder, bandwidth_observer,
remote_bitrate_estimator)) {
bitrate_observer)) {
vie_encoder = NULL;
}
}
@ -201,9 +202,7 @@ int ViEChannelManager::DeleteChannel(int channel_id) {
group = FindGroup(channel_id);
group->SetChannelRembStatus(channel_id, false, false, vie_channel,
vie_encoder);
unsigned int ssrc = 0;
vie_channel->GetRemoteSSRC(ssrc);
group->RemoveChannel(channel_id, ssrc);
group->RemoveChannel(channel_id);
// Check if other channels are using the same encoder.
if (ChannelUsingViEEncoder(channel_id)) {
@ -327,7 +326,7 @@ bool ViEChannelManager::CreateChannelObject(
int channel_id,
ViEEncoder* vie_encoder,
RtcpBandwidthObserver* bandwidth_observer,
RemoteBitrateEstimator* remote_bitrate_estimator) {
RtpRemoteBitrateObserver* bitrate_observer) {
// Register the channel at the encoder.
RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
@ -336,7 +335,7 @@ bool ViEChannelManager::CreateChannelObject(
*module_process_thread_,
vie_encoder,
bandwidth_observer,
remote_bitrate_estimator,
bitrate_observer,
send_rtp_rtcp_module);
if (vie_channel->Init() != 0) {
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),

2
src/video_engine/vie_channel_manager.h
View File

@ -78,7 +78,7 @@ class ViEChannelManager: private ViEManagerBase {
// protected.
bool CreateChannelObject(int channel_id, ViEEncoder* vie_encoder,
RtcpBandwidthObserver* bandwidth_observer,
RemoteBitrateEstimator* remote_bitrate_estimator);
RtpRemoteBitrateObserver* bitrate_observer);
// Used by ViEChannelScoped, forcing a manager user to use scoped.
// Returns a pointer to the channel with id 'channelId'.

3
src/video_engine/vie_remb.h
View File

@ -22,7 +22,6 @@
#include <map>
#include "modules/interface/module.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "system_wrappers/interface/scoped_ptr.h"
@ -32,7 +31,7 @@ class CriticalSectionWrapper;
class ProcessThread;
class RtpRtcp;
class VieRemb : public RemoteBitrateObserver, public Module {
class VieRemb : public RtpRemoteBitrateObserver, public Module {
public:
VieRemb(ProcessThread* process_thread);
~VieRemb();