New attempt to cleanup TMMBR.
Review URL: https://webrtc-codereview.appspot.com/472007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1990 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/source/bandwidth_management.cc b/src/modules/rtp_rtcp/source/bandwidth_management.cc
index a14a2ef..b1402da 100644
--- a/src/modules/rtp_rtcp/source/bandwidth_management.cc
+++ b/src/modules/rtp_rtcp/source/bandwidth_management.cc
@@ -31,8 +31,6 @@
_last_fraction_loss(0),
_last_round_trip_time(0),
_bwEstimateIncoming(0),
- _smoothedFractionLostQ4(-1), // indicate uninitialized
- _sFLFactorQ4(14), // 0.875 in Q4
_timeLastIncrease(0)
{
}
@@ -269,20 +267,7 @@
newBitRate += 1000;
}
- // Calculate smoothed loss number
- if (_smoothedFractionLostQ4 < 0)
- {
- // startup
- _smoothedFractionLostQ4 = static_cast<WebRtc_UWord16>(packetLoss);
- }
- else
- {
- _smoothedFractionLostQ4 = ((_sFLFactorQ4 * _smoothedFractionLostQ4 + 8) >> 4) // Q4*Q4 = Q8; down to Q4 again with proper rounding
- + (16 - _sFLFactorQ4) * static_cast<WebRtc_UWord16>(packetLoss); // Q4 * Q0 = Q4
- }
-
// Calculate what rate TFRC would apply in this situation
- //WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, _smoothedFractionLostQ4 >> 4); // scale loss to Q0 (back to [0, 255])
WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, packetLoss); // scale loss to Q0 (back to [0, 255])
if (reducing &&
@@ -290,7 +275,7 @@
static_cast<WebRtc_UWord32>(tfrcRate) > newBitRate)
{
// do not reduce further if rate is below TFRC rate
- newBitRate = _bitRate;
+ newBitRate = tfrcRate;
}
if (_bwEstimateIncoming > 0 && newBitRate > _bwEstimateIncoming)
diff --git a/src/modules/rtp_rtcp/source/bandwidth_management.h b/src/modules/rtp_rtcp/source/bandwidth_management.h
index 2516b86..b743160 100644
--- a/src/modules/rtp_rtcp/source/bandwidth_management.h
+++ b/src/modules/rtp_rtcp/source/bandwidth_management.h
@@ -85,8 +85,6 @@
// bandwidth estimate
WebRtc_UWord32 _bwEstimateIncoming;
- WebRtc_Word16 _smoothedFractionLostQ4;
- WebRtc_Word16 _sFLFactorQ4; // forgetting factor for _smoothedFractionLostQ4
WebRtc_Word64 _timeLastIncrease;
};
} // namespace webrtc
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
index f4eca71..d65afdf 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -27,28 +27,28 @@
using namespace RTCPUtility;
using namespace RTCPHelp;
-RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id,
- RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner) :
- _id(id),
- _clock(*clock),
- _method(kRtcpOff),
- _lastReceived(0),
- _rtpRtcp(*owner),
- _criticalSectionFeedbacks(CriticalSectionWrapper::CreateCriticalSection()),
- _cbRtcpFeedback(NULL),
- _cbVideoFeedback(NULL),
- _criticalSectionRTCPReceiver(
- CriticalSectionWrapper::CreateCriticalSection()),
- _SSRC(0),
- _remoteSSRC(0),
- _remoteSenderInfo(),
- _lastReceivedSRNTPsecs(0),
- _lastReceivedSRNTPfrac(0),
- _receivedInfoMap(),
- _packetTimeOutMS(0),
- _rtt(0)
-{
+RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock,
+ ModuleRtpRtcpImpl* owner)
+ : TMMBRHelp(),
+ _id(id),
+ _clock(*clock),
+ _method(kRtcpOff),
+ _lastReceived(0),
+ _rtpRtcp(*owner),
+ _criticalSectionFeedbacks(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _cbRtcpFeedback(NULL),
+ _cbVideoFeedback(NULL),
+ _criticalSectionRTCPReceiver(
+ CriticalSectionWrapper::CreateCriticalSection()),
+ _SSRC(0),
+ _remoteSSRC(0),
+ _remoteSenderInfo(),
+ _lastReceivedSRNTPsecs(0),
+ _lastReceivedSRNTPfrac(0),
+ _receivedInfoMap(),
+ _packetTimeOutMS(0),
+ _rtt(0) {
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
@@ -1224,15 +1224,58 @@
}
}
-void
-RTCPReceiver::OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64 pitureID) const
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
+void RTCPReceiver::OnReceivedReferencePictureSelectionIndication(
+ const WebRtc_UWord64 pitureID) const {
+ CriticalSectionScoped lock(_criticalSectionFeedbacks);
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnRPSIReceived(_id, pitureID);
- }
+ if (_cbRtcpFeedback) {
+ _cbRtcpFeedback->OnRPSIReceived(_id, pitureID);
+ }
+}
+
+WebRtc_Word32 RTCPReceiver::UpdateTMMBR() {
+ WebRtc_Word32 numBoundingSet = 0;
+ WebRtc_UWord32 minBitrateKbit = 0;
+ WebRtc_UWord32 maxBitrateKbit = 0;
+ WebRtc_UWord32 accNumCandidates = 0;
+
+ WebRtc_Word32 size = TMMBRReceived(0, 0, NULL);
+ if (size > 0) {
+ TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
+ // Get candidate set from receiver.
+ accNumCandidates = TMMBRReceived(size, accNumCandidates, candidateSet);
+ } else {
+ // Candidate set empty.
+ VerifyAndAllocateCandidateSet(0); // resets candidate set
+ }
+ // Find bounding set
+ TMMBRSet* boundingSet = NULL;
+ numBoundingSet = FindTMMBRBoundingSet(boundingSet);
+ if (numBoundingSet == -1) {
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "Failed to find TMMBR bounding set.");
+ return -1;
+ }
+ // Set bounding set
+ // Inform remote clients about the new bandwidth
+ // inform the remote client
+ _rtpRtcp.SetTMMBN(boundingSet);
+
+ // might trigger a TMMBN
+ if (numBoundingSet == 0) {
+ // owner of max bitrate request has timed out
+ // empty bounding set has been sent
+ return 0;
+ }
+ // Get net bitrate from bounding set depending on sent packet rate
+ if (CalcMinBitRate(&minBitrateKbit)) {
+ // we have a new bandwidth estimate on this channel
+ _rtpRtcp.OnReceivedBandwidthEstimateUpdate((WebRtc_UWord16)minBitrateKbit);
+ WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
+ "Set TMMBR request min:%d kbps max:%d kbps, channel: %d",
+ minBitrateKbit, maxBitrateKbit, _id);
+ }
+ return 0;
}
// Holding no Critical section
@@ -1247,7 +1290,7 @@
"SIG [RTCP] Incoming TMMBR to id:%d", _id);
// Might trigger a OnReceivedBandwidthEstimateUpdate.
- _rtpRtcp.OnReceivedTMMBR();
+ UpdateTMMBR();
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
{
@@ -1387,18 +1430,6 @@
}
}
-void
-RTCPReceiver::UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit)
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnTMMBRReceived(_id, bwEstimateKbit);
- }
-
-}
-
WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
char cName[RTCP_CNAME_SIZE]) const {
if (cName == NULL) {
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.h b/src/modules/rtp_rtcp/source/rtcp_receiver.h
index eb1b658..1e67b44 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -19,11 +19,12 @@
#include "rtcp_utility.h"
#include "rtp_rtcp_defines.h"
#include "rtcp_receiver_help.h"
+#include "tmmbr_help.h"
namespace webrtc {
class ModuleRtpRtcpImpl;
-class RTCPReceiver
+class RTCPReceiver : public TMMBRHelp
{
public:
RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock,
@@ -97,10 +98,9 @@
bool UpdateRTCPReceiveInformationTimers();
- void UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit);
+ WebRtc_Word32 BoundingSet(bool &tmmbrOwner, TMMBRSet*& boundingSetRec);
- WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
- TMMBRSet*& boundingSetRec);
+ WebRtc_Word32 UpdateTMMBR();
WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
void PacketTimeout();
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.cc b/src/modules/rtp_rtcp/source/rtcp_sender.cc
index 58e849f..78af109 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -68,7 +68,7 @@
_rembBitrate(0),
_bitrate_observer(NULL),
- _tmmbrHelp(audio),
+ _tmmbrHelp(),
_tmmbr_Send(0),
_packetOH_Send(0),
_remoteRateControl(),
@@ -2165,20 +2165,6 @@
return -1;
}
-WebRtc_Word32
-RTCPSender::RequestTMMBR(WebRtc_UWord32 estimatedBW, WebRtc_UWord32 packetOH)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if(_TMMBR)
- {
- _tmmbr_Send = estimatedBW;
- _packetOH_Send = packetOH;
-
- return 0;
- }
- return -1;
-}
-
RateControlRegion
RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse)
{
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.h b/src/modules/rtp_rtcp/source/rtcp_sender.h
index 966b64f..38a264e 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.h
@@ -103,9 +103,6 @@
WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
const WebRtc_UWord32 maxBitrateKbit);
- WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH);
-
/*
* Extended jitter report
*/
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 5fb591e..84f3eef 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -82,7 +82,6 @@
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
const bool audio,
RtpRtcpClock* clock):
- TMMBRHelp(audio),
_rtpSender(id, audio, clock),
_rtpReceiver(id, audio, clock, this),
_rtcpSender(id, audio, clock, this),
@@ -266,7 +265,7 @@
CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
// we use two locks for protecting _childModules one
// (_criticalSectionModulePtrsFeedback) for incoming
- // messages (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs
+ // messages (BitrateSent) and _criticalSectionModulePtrs
// for all outgoing messages sending packets etc
_childModules.push_back((ModuleRtpRtcpImpl*)module);
}
@@ -415,19 +414,21 @@
// default module or no RTCP received yet.
max_rtt = kDefaultRtt;
}
- if (REMB() && _rtcpSender.ValidBitrateEstimate()) {
- unsigned int 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.UpdateRemoteBitrateEstimate(target_bitrate);
- } else if (TMMBR()) {
- _rtcpSender.CalculateNewTargetBitrate(max_rtt);
+ }
}
_rtcpSender.SendRTCP(kRtcpReport);
}
if (UpdateRTCPReceiveInformationTimers()) {
// a receiver has timed out
- UpdateTMMBR();
+ _rtcpReceiver.UpdateTMMBR();
}
return 0;
}
@@ -1670,29 +1671,13 @@
return _rtcpSender.SetTMMBRStatus(enable);
}
-WebRtc_Word32 ModuleRtpRtcpImpl::TMMBRReceived(
- const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TMMBRReceived()");
-
- return _rtcpReceiver.TMMBRReceived(size, accNumCandidates, candidateSet);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit) {
+WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* boundingSet) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetTMMBN()");
+ WebRtc_UWord32 maxBitrateKbit = _rtpSender.MaxConfiguredBitrateVideo() / 1000;
return _rtcpSender.SetTMMBN(boundingSet, maxBitrateKbit);
}
-WebRtc_Word32 ModuleRtpRtcpImpl::RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RequestTMMBR()");
-
- return _rtcpSender.RequestTMMBR(estimatedBW, packetOH);
-}
-
/*
* (NACK) Negative acknowledgement
*/
@@ -2299,48 +2284,6 @@
}
}
-// our local BW estimate is updated
-void ModuleRtpRtcpImpl::OnBandwidthEstimateUpdate(
- WebRtc_UWord16 bandWidthKbit) {
-
- WebRtc_UWord32 maxBitrateKbit = _rtpReceiver.MaxConfiguredBitrate() / 1000;
- if (maxBitrateKbit) {
- // the app has set a max bitrate
- if (maxBitrateKbit < bandWidthKbit) {
- // cap TMMBR at max configured bitrate
- bandWidthKbit = (WebRtc_UWord16)maxBitrateKbit;
- }
- }
- if (_rtcpSender.TMMBR()) {
- /* Maximum total media bit rate:
- The upper limit on total media bit rate for a given media
- stream at a particular receiver and for its selected protocol
- layer. Note that this value cannot be measured on the
- received media stream. Instead, it needs to be calculated or
- determined through other means, such as quality of service
- (QoS) negotiations or local resource limitations. Also note
- that this value is an average (on a timescale that is
- reasonable for the application) and that it may be different
- from the instantaneous bit rate seen by packets in the media
- stream.
- */
- /* Overhead:
- All protocol header information required to convey a packet
- with media data from sender to receiver, from the application
- layer down to a pre-defined protocol level (for example, down
- to, and including, the IP header). Overhead may include, for
- example, IP, UDP, and RTP headers, any layer 2 headers, any
- Contributing Sources (CSRCs), RTP padding, and RTP header
- extensions. Overhead excludes any RTP payload headers and the
- payload itself.
- */
- _rtpReceiver.PacketOHReceived();
-
- // call RequestTMMBR when our localy created estimate changes
- _rtcpSender.RequestTMMBR(bandWidthKbit, 0);
- }
-}
-
RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
const RateControlInput& rateControlInput) {
@@ -2506,34 +2449,32 @@
void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
const WebRtc_UWord16 bwEstimateKbit) {
-
// We received a TMMBR
+ if (_audio) {
+ return;
+ }
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
ProcessDefaultModuleBandwidth();
return;
}
- if (_audio) {
- _rtcpReceiver.UpdateBandwidthEstimate(bwEstimateKbit);
- } else {
- WebRtc_UWord32 newBitrate = 0;
- WebRtc_UWord8 fractionLost = 0;
- WebRtc_UWord16 roundTripTime = 0;
- if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
- &newBitrate,
- &fractionLost,
- &roundTripTime) == 0) {
- if (!_defaultModule) {
- // No default module check if we should trigger OnNetworkChanged
- // via video callback
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- fractionLost,
- roundTripTime);
- }
- if (newBitrate > 0) {
- // update bitrate
- _rtpSender.SetTargetSendBitrate(newBitrate);
- }
+ WebRtc_UWord32 newBitrate = 0;
+ WebRtc_UWord8 fractionLost = 0;
+ WebRtc_UWord16 roundTripTime = 0;
+ if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
+ &newBitrate,
+ &fractionLost,
+ &roundTripTime) == 0) {
+ if (!_defaultModule) {
+ // No default module check if we should trigger OnNetworkChanged
+ // via video callback
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ fractionLost,
+ roundTripTime);
+ }
+ if (newBitrate > 0) {
+ // update bitrate
+ _rtpSender.SetTargetSendBitrate(newBitrate);
}
}
if (_defaultModule) {
@@ -2768,88 +2709,16 @@
return 0;
}
-void ModuleRtpRtcpImpl::OnReceivedTMMBR() {
- // we received a TMMBR in a RTCP packet
- // answer with a TMMBN
- UpdateTMMBR();
-}
-
bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
// if this returns true this channel has timed out
// periodically check if this is true and if so call UpdateTMMBR
return _rtcpReceiver.UpdateRTCPReceiveInformationTimers();
}
-WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
- WebRtc_Word32 numBoundingSet = 0;
- WebRtc_Word32 newBitrates = 0;
- WebRtc_UWord32 minBitrateKbit = 0;
- WebRtc_UWord32 maxBitrateKbit = 0;
- WebRtc_UWord32 accNumCandidates = 0;
-
- if (!_childModules.empty()) {
- // Default module should not handle this
- return -1;
- }
-
- WebRtc_Word32 size = _rtcpReceiver.TMMBRReceived(0, 0, NULL);
- if (size > 0) {
- TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
- // get candidate set from receiver
- accNumCandidates = _rtcpReceiver.TMMBRReceived(size,
- accNumCandidates,
- candidateSet);
- } else {
- // candidate set empty
- VerifyAndAllocateCandidateSet(0); // resets candidate set
- }
- // Find bounding set
- TMMBRSet* boundingSet = NULL;
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- if (numBoundingSet == -1) {
- WEBRTC_TRACE(kTraceWarning,
- kTraceRtpRtcp,
- _id,
- "Failed to find TMMBR bounding set.");
- return -1;
- }
- // Set bounding set
- // Inform remote clients about the new bandwidth
- // inform the remote client
- _rtcpSender.SetTMMBN(boundingSet,
- _rtpSender.MaxConfiguredBitrateVideo() / 1000);
- // might trigger a TMMBN
- if (numBoundingSet == 0) {
- // owner of max bitrate request has timed out
- // empty bounding set has been sent
- return 0;
- }
- // Get net bitrate from bounding set depending on sent packet rate
- newBitrates = CalcMinMaxBitRate(_rtpSender.PacketRate(),
- (WebRtc_UWord32)numBoundingSet,
- minBitrateKbit,
- maxBitrateKbit);
-
- // no critsect when calling out to "unknown" code
- if (newBitrates == 0) {
- // we have new bitrate
- // Set new max bitrate
- // we have a new bandwidth estimate on this channel
- OnReceivedBandwidthEstimateUpdate((WebRtc_UWord16)minBitrateKbit);
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- _id,
- "Set TMMBR request min:%d kbps max:%d kbps, channel: %d",
- minBitrateKbit, maxBitrateKbit, _id);
- }
- return 0;
-}
-
// called from RTCPsender
WebRtc_Word32 ModuleRtpRtcpImpl::BoundingSet(bool& tmmbrOwner,
TMMBRSet*& boundingSet) {
- return _rtcpReceiver.BoundingSet(tmmbrOwner,
- boundingSet);
+ return _rtcpReceiver.BoundingSet(tmmbrOwner, boundingSet);
}
void ModuleRtpRtcpImpl::SendKeyFrame() {
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 9236061..0dca764 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -26,7 +26,7 @@
namespace webrtc {
-class ModuleRtpRtcpImpl : public RtpRtcp, private TMMBRHelp
+class ModuleRtpRtcpImpl : public RtpRtcp
{
public:
ModuleRtpRtcpImpl(const WebRtc_Word32 id,
@@ -341,15 +341,7 @@
virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
- virtual WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const;
-
- virtual WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit);
-
- virtual WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH);
+ WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet);
virtual WebRtc_UWord16 MaxPayloadLength() const;
@@ -496,8 +488,6 @@
// good state of RTP receiver inform sender
virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
- virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit);
-
void OnReceivedNTP() ;
// bw estimation
@@ -547,8 +537,6 @@
// Get remote SequenceNumber
WebRtc_UWord16 RemoteSequenceNumber() const;
- WebRtc_Word32 UpdateTMMBR();
-
// only for internal testing
WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
diff --git a/src/modules/rtp_rtcp/source/tmmbr_help.cc b/src/modules/rtp_rtcp/source/tmmbr_help.cc
index cf34e08..6fba710 100644
--- a/src/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/src/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -10,6 +10,7 @@
#include "tmmbr_help.h"
+#include <limits>
#include "rtp_rtcp_config.h"
namespace webrtc {
@@ -61,24 +62,21 @@
lengthOfSet = 0;
}
-TMMBRHelp::TMMBRHelp(const bool audio) :
- _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
- _audio(audio),
- _candidateSet(),
- _boundingSet(),
- _boundingSetToSend(),
- _ptrIntersectionBoundingSet(NULL),
- _ptrMaxPRBoundingSet(NULL)
-{
+TMMBRHelp::TMMBRHelp()
+ : _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
+ _candidateSet(),
+ _boundingSet(),
+ _boundingSetToSend(),
+ _ptrIntersectionBoundingSet(NULL),
+ _ptrMaxPRBoundingSet(NULL) {
}
-TMMBRHelp::~TMMBRHelp()
-{
- delete [] _ptrIntersectionBoundingSet;
- delete [] _ptrMaxPRBoundingSet;
- _ptrIntersectionBoundingSet = 0;
- _ptrMaxPRBoundingSet = 0;
- delete _criticalSection;
+TMMBRHelp::~TMMBRHelp() {
+ delete [] _ptrIntersectionBoundingSet;
+ delete [] _ptrMaxPRBoundingSet;
+ _ptrIntersectionBoundingSet = 0;
+ _ptrMaxPRBoundingSet = 0;
+ delete _criticalSection;
}
TMMBRSet*
@@ -101,10 +99,8 @@
return &_boundingSet;
}
-TMMBRSet*
-TMMBRHelp::BoundingSet()
-{
- return &_boundingSet;
+TMMBRSet* TMMBRHelp::BoundingSet() {
+ return &_boundingSet;
}
WebRtc_Word32
@@ -413,95 +409,39 @@
return numBoundingSet;
}
-bool
-TMMBRHelp::IsOwner(const WebRtc_UWord32 ssrc,
- const WebRtc_UWord32 length) const
-{
- CriticalSectionScoped lock(_criticalSection);
+bool TMMBRHelp::IsOwner(const WebRtc_UWord32 ssrc,
+ const WebRtc_UWord32 length) const {
+ CriticalSectionScoped lock(_criticalSection);
- if (length == 0)
- {
- // empty bounding set
- return false;
- }
-
- for(WebRtc_UWord32 i = 0; (i < length) && (i < _boundingSet.sizeOfSet); ++i)
- {
- if(_boundingSet.ptrSsrcSet[i] == ssrc)
- {
- return true;
- }
- }
+ if (length == 0) {
+ // Empty bounding set.
return false;
+ }
+ for(WebRtc_UWord32 i = 0; (i < length) && (i < _boundingSet.sizeOfSet); ++i) {
+ if(_boundingSet.ptrSsrcSet[i] == ssrc) {
+ return true;
+ }
+ }
+ return false;
}
-WebRtc_Word32
-TMMBRHelp::CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
- const WebRtc_UWord32 lengthOfBoundingSet,
- WebRtc_UWord32& minBitrateKbit,
- WebRtc_UWord32& maxBitrateKbit) const
-{
- CriticalSectionScoped lock(_criticalSection);
+bool TMMBRHelp::CalcMinBitRate( WebRtc_UWord32* minBitrateKbit) const {
+ CriticalSectionScoped lock(_criticalSection);
- if (lengthOfBoundingSet <= 0 || _candidateSet.sizeOfSet == 0)
- {
- // empty bounding set
- return -1;
+ if (_candidateSet.sizeOfSet == 0) {
+ // Empty bounding set.
+ return false;
+ }
+ *minBitrateKbit = std::numeric_limits<uint32_t>::max();
+
+ for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; ++i) {
+ WebRtc_UWord32 curNetBitRateKbit = _candidateSet.ptrTmmbrSet[i];
+ if (curNetBitRateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE) {
+ curNetBitRateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
}
-
- minBitrateKbit = 0xFFFFFFFF;
- maxBitrateKbit = 0;
-
- for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; ++i)
- {
- if (_candidateSet.ptrTmmbrSet[i])
- {
- WebRtc_Word32 curNetBitRate = static_cast<WebRtc_Word32>((_candidateSet.ptrTmmbrSet[i]*1000.0
- - (totalPacketRate * (_candidateSet.ptrPacketOHSet[i] << 3)))/1000 + 0.5);
-
- if (curNetBitRate < 0)
- {
- // could be negative for a large packet rate
- if(_audio)
- {
- curNetBitRate = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }else
- {
- curNetBitRate = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- }
- minBitrateKbit = (WebRtc_UWord32(curNetBitRate) < minBitrateKbit) ? curNetBitRate : minBitrateKbit;
- }
- }
- maxBitrateKbit = minBitrateKbit;
-
- if (maxBitrateKbit == 0 || maxBitrateKbit < minBitrateKbit)
- {
- return -1;
- }
-
- if(_audio)
- {
- if (minBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
- {
- minBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }
- if (maxBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
- {
- maxBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }
- }else
- {
- if (minBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
- {
- minBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- if (maxBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
- {
- maxBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- }
-
- return 0;
+ *minBitrateKbit = curNetBitRateKbit < *minBitrateKbit ?
+ curNetBitRateKbit : *minBitrateKbit;
+ }
+ return true;
}
} // namespace webrtc
diff --git a/src/modules/rtp_rtcp/source/tmmbr_help.h b/src/modules/rtp_rtcp/source/tmmbr_help.h
index 35704fe..6b3cd97 100644
--- a/src/modules/rtp_rtcp/source/tmmbr_help.h
+++ b/src/modules/rtp_rtcp/source/tmmbr_help.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -38,7 +38,7 @@
class TMMBRHelp
{
public:
- TMMBRHelp(const bool audio);
+ TMMBRHelp();
virtual ~TMMBRHelp();
TMMBRSet* BoundingSet(); // used for debuging
@@ -47,16 +47,13 @@
TMMBRSet* VerifyAndAllocateCandidateSet(const WebRtc_UWord32 minimumSize);
WebRtc_Word32 FindTMMBRBoundingSet(TMMBRSet*& boundingSet);
- WebRtc_Word32 SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend,
- const WebRtc_UWord32 maxBitrateKbit);
+ WebRtc_Word32 SetTMMBRBoundingSetToSend(
+ const TMMBRSet* boundingSetToSend,
+ const WebRtc_UWord32 maxBitrateKbit);
- bool IsOwner(const WebRtc_UWord32 ssrc,
- const WebRtc_UWord32 length) const;
+ bool IsOwner(const WebRtc_UWord32 ssrc, const WebRtc_UWord32 length) const;
- WebRtc_Word32 CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
- const WebRtc_UWord32 lengthOfBoundingSet,
- WebRtc_UWord32& minBitrateKbit,
- WebRtc_UWord32& maxBitrateKbit) const;
+ bool CalcMinBitRate(WebRtc_UWord32* minBitrateKbit) const;
protected:
TMMBRSet* VerifyAndAllocateBoundingSet(WebRtc_UWord32 minimumSize);
@@ -66,7 +63,6 @@
private:
CriticalSectionWrapper* _criticalSection;
- const bool _audio;
TMMBRSet _candidateSet;
TMMBRSet _boundingSet;
TMMBRSet _boundingSetToSend;