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Only trigger one call to OnNetworkChanged for each incoming RTCP packet.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
2011-11-24 15:22:33 +00:00
parent 471e83e592
commit 26b9777e62
3 changed files with 41 additions and 20 deletions
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9
src/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -1278,10 +1278,17 @@ RTCPReceiver::TriggerCallbacksFromRTCPPacket(RTCPPacketInformation& rtcpPacketIn
{ {
if(rtcpPacketInformation.reportBlock) if(rtcpPacketInformation.reportBlock)
{ {
// We only want to trigger one OnNetworkChanged callback per RTCP
// packet. The callback is triggered by a SR, RR and TMMBR, so we
// don't want to trigger one from here if the packet also contains a
// TMMBR block.
bool triggerCallback =
!(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr);
_rtpRtcp.OnPacketLossStatisticsUpdate( _rtpRtcp.OnPacketLossStatisticsUpdate(
rtcpPacketInformation.fractionLost, rtcpPacketInformation.fractionLost,
rtcpPacketInformation.roundTripTime, rtcpPacketInformation.roundTripTime,
rtcpPacketInformation.lastReceivedExtendedHighSeqNum); rtcpPacketInformation.lastReceivedExtendedHighSeqNum,
triggerCallback);
} }
} }
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)

47
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -2633,7 +2633,7 @@ void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
// We received a TMMBR // We received a TMMBR
const bool defaultInstance(_childModules.empty() ? false : true); const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) { if (defaultInstance) {
ProcessDefaultModuleBandwidth(); ProcessDefaultModuleBandwidth(true);
return; return;
} }
if (_audio) { if (_audio) {
@ -2674,7 +2674,8 @@ void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate( void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost, const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime, const WebRtc_UWord16 roundTripTime,
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum) { const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
bool triggerOnNetworkChanged) {
const bool defaultInstance(_childModules.empty() ? false : true); const bool defaultInstance(_childModules.empty() ? false : true);
if (!defaultInstance) { if (!defaultInstance) {
@ -2709,18 +2710,22 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
_defaultModule->OnPacketLossStatisticsUpdate( _defaultModule->OnPacketLossStatisticsUpdate(
loss, // send in the filtered loss loss, // send in the filtered loss
roundTripTime, roundTripTime,
lastReceivedExtendedHighSeqNum); lastReceivedExtendedHighSeqNum,
triggerOnNetworkChanged);
} }
return; return;
} }
// No default module check if we should trigger OnNetworkChanged // No default module check if we should trigger OnNetworkChanged
// via video callback // via video callback
_rtpReceiver.UpdateBandwidthManagement(newBitrate, if (triggerOnNetworkChanged)
fractionLost, {
roundTripTime); _rtpReceiver.UpdateBandwidthManagement(newBitrate,
fractionLost,
roundTripTime);
}
} else { } else {
if (!_simulcast) { if (!_simulcast) {
ProcessDefaultModuleBandwidth(); ProcessDefaultModuleBandwidth(triggerOnNetworkChanged);
} else { } else {
// default and simulcast // default and simulcast
WebRtc_UWord32 newBitrate = 0; WebRtc_UWord32 newBitrate = 0;
@ -2740,9 +2745,12 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
_rtpSender.SetTargetSendBitrate(newBitrate); _rtpSender.SetTargetSendBitrate(newBitrate);
// check if we should trigger OnNetworkChanged // check if we should trigger OnNetworkChanged
// via video callback // via video callback
_rtpReceiver.UpdateBandwidthManagement(newBitrate, if (triggerOnNetworkChanged)
loss, {
roundTripTime); _rtpReceiver.UpdateBandwidthManagement(newBitrate,
loss,
roundTripTime);
}
// sanity // sanity
if (_sendVideoCodec.codecType == kVideoCodecUnknown) { if (_sendVideoCodec.codecType == kVideoCodecUnknown) {
return; return;
@ -2779,7 +2787,9 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
} }
} }
void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() { void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth(
bool triggerOnNetworkChanged) {
WebRtc_UWord32 minBitrateBps = 0xffffffff; WebRtc_UWord32 minBitrateBps = 0xffffffff;
WebRtc_UWord32 maxBitrateBps = 0; WebRtc_UWord32 maxBitrateBps = 0;
WebRtc_UWord32 count = 0; WebRtc_UWord32 count = 0;
@ -2833,12 +2843,15 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
return; return;
} }
_bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0); _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
// Update default module bitrate. Don't care about min max.
// Check if we should trigger OnNetworkChanged via video callback if (triggerOnNetworkChanged) {
WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count); // Update default module bitrate. Don't care about min max.
_rtpReceiver.UpdateBandwidthManagement(minBitrateBps, // Check if we should trigger OnNetworkChanged via video callback
fractionLostAvg , WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
maxRoundTripTime); _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
fractionLostAvg ,
maxRoundTripTime);
}
} }
void ModuleRtpRtcpImpl::OnRequestSendReport() { void ModuleRtpRtcpImpl::OnRequestSendReport() {

5
src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -481,7 +481,8 @@ public:
void OnPacketLossStatisticsUpdate( void OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost, const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime, const WebRtc_UWord16 roundTripTime,
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum); const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
bool triggerOnNetworkChanged);
void OnReceivedTMMBR(); void OnReceivedTMMBR();
@ -532,7 +533,7 @@ protected:
RTCPReceiver _rtcpReceiver; RTCPReceiver _rtcpReceiver;
private: private:
void SendKeyFrame(); void SendKeyFrame();
void ProcessDefaultModuleBandwidth(); void ProcessDefaultModuleBandwidth(bool triggerOnNetworkChanged);
WebRtc_Word32 _id; WebRtc_Word32 _id;
const bool _audio; const bool _audio;