Only trigger one call to OnNetworkChanged for each incoming RTCP packet.
Review URL: http://webrtc-codereview.appspot.com/289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
index 4132367..8b4686c 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -1278,10 +1278,17 @@
{
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(
rtcpPacketInformation.fractionLost,
rtcpPacketInformation.roundTripTime,
- rtcpPacketInformation.lastReceivedExtendedHighSeqNum);
+ rtcpPacketInformation.lastReceivedExtendedHighSeqNum,
+ triggerCallback);
}
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index ffc31b3..cc08583 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -2633,7 +2633,7 @@
// We received a TMMBR
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
- ProcessDefaultModuleBandwidth();
+ ProcessDefaultModuleBandwidth(true);
return;
}
if (_audio) {
@@ -2674,7 +2674,8 @@
void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime,
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum) {
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
+ bool triggerOnNetworkChanged) {
const bool defaultInstance(_childModules.empty() ? false : true);
if (!defaultInstance) {
@@ -2709,18 +2710,22 @@
_defaultModule->OnPacketLossStatisticsUpdate(
loss, // send in the filtered loss
roundTripTime,
- lastReceivedExtendedHighSeqNum);
+ lastReceivedExtendedHighSeqNum,
+ triggerOnNetworkChanged);
}
return;
}
// No default module check if we should trigger OnNetworkChanged
// via video callback
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- fractionLost,
- roundTripTime);
+ if (triggerOnNetworkChanged)
+ {
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ fractionLost,
+ roundTripTime);
+ }
} else {
if (!_simulcast) {
- ProcessDefaultModuleBandwidth();
+ ProcessDefaultModuleBandwidth(triggerOnNetworkChanged);
} else {
// default and simulcast
WebRtc_UWord32 newBitrate = 0;
@@ -2740,9 +2745,12 @@
_rtpSender.SetTargetSendBitrate(newBitrate);
// check if we should trigger OnNetworkChanged
// via video callback
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- loss,
- roundTripTime);
+ if (triggerOnNetworkChanged)
+ {
+ _rtpReceiver.UpdateBandwidthManagement(newBitrate,
+ loss,
+ roundTripTime);
+ }
// sanity
if (_sendVideoCodec.codecType == kVideoCodecUnknown) {
return;
@@ -2779,7 +2787,9 @@
}
}
-void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
+void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth(
+ bool triggerOnNetworkChanged) {
+
WebRtc_UWord32 minBitrateBps = 0xffffffff;
WebRtc_UWord32 maxBitrateBps = 0;
WebRtc_UWord32 count = 0;
@@ -2833,12 +2843,15 @@
return;
}
_bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
- // Update default module bitrate. Don't care about min max.
- // Check if we should trigger OnNetworkChanged via video callback
- WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
- _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
- fractionLostAvg ,
- maxRoundTripTime);
+
+ if (triggerOnNetworkChanged) {
+ // Update default module bitrate. Don't care about min max.
+ // Check if we should trigger OnNetworkChanged via video callback
+ WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
+ _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
+ fractionLostAvg ,
+ maxRoundTripTime);
+ }
}
void ModuleRtpRtcpImpl::OnRequestSendReport() {
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 462a665..560a60f 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -481,7 +481,8 @@
void OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime,
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum);
+ const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
+ bool triggerOnNetworkChanged);
void OnReceivedTMMBR();
@@ -532,7 +533,7 @@
RTCPReceiver _rtcpReceiver;
private:
void SendKeyFrame();
- void ProcessDefaultModuleBandwidth();
+ void ProcessDefaultModuleBandwidth(bool triggerOnNetworkChanged);
WebRtc_Word32 _id;
const bool _audio;
