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Bugfix OnNetworkChanged not triggered for RTCP compund messages if TMMBR is higher than last value.

Browse Source 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/342001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1344 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-01-05 08:40:56 +00:00
parent 401045a0d4
commit 3aa25de346
3 changed files with 51 additions and 69 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

67
src/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -1301,29 +1301,37 @@ RTCPReceiver::OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64
}
// Holding no Critical section
void
RTCPReceiver::TriggerCallbacksFromRTCPPacket(RTCPPacketInformation& rtcpPacketInformation)
void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
RTCPPacketInformation& rtcpPacketInformation)
{
// callback if SR or RR
// Process TMMBR and REMB first to avoid multiple callbacks
// to OnNetworkChanged.
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
"SIG [RTCP] Incoming TMMBR to id:%d", _id);
// Might trigger a OnReceivedBandwidthEstimateUpdate.
_rtpRtcp.OnReceivedTMMBR();
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
"SIG [RTCP] Incoming REMB to id:%d", _id);
// We need to bounce this to the default channel.
_rtpRtcp.OnReceivedEstimatedMaxBitrate(
rtcpPacketInformation.receiverEstimatedMaxBitrate);
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr)
{
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, REMB or TMMBR, so
// we don't want to trigger one from here if the packet also
// contains a REMB or TMMBR block.
bool triggerCallback = true;
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb ||
rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr) {
triggerCallback = false;
}
_rtpRtcp.OnPacketLossStatisticsUpdate(
rtcpPacketInformation.fractionLost,
rtcpPacketInformation.roundTripTime,
rtcpPacketInformation.lastReceivedExtendedHighSeqNum,
triggerCallback);
rtcpPacketInformation.lastReceivedExtendedHighSeqNum);
}
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
@ -1338,27 +1346,24 @@ RTCPReceiver::TriggerCallbacksFromRTCPPacket(RTCPPacketInformation& rtcpPacketIn
{
if (rtcpPacketInformation.nackSequenceNumbersLength > 0)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming NACK to id:%d", _id);
_rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbersLength,
rtcpPacketInformation.nackSequenceNumbers);
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
"SIG [RTCP] Incoming NACK to id:%d", _id);
_rtpRtcp.OnReceivedNACK(
rtcpPacketInformation.nackSequenceNumbersLength,
rtcpPacketInformation.nackSequenceNumbers);
}
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming TMMBR to id:%d", _id);
// might trigger a OnReceivedBandwidthEstimateUpdate
_rtpRtcp.OnReceivedTMMBR();
}
if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir))
{
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli)
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming PLI to id:%d", _id);
}else
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
"SIG [RTCP] Incoming PLI to id:%d", _id);
} else
{
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming FIR to id:%d", _id);
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
"SIG [RTCP] Incoming FIR to id:%d", _id);
}
_rtpRtcp.OnReceivedIntraFrameRequest(&_rtpRtcp);
}
@ -1368,12 +1373,6 @@ RTCPReceiver::TriggerCallbacksFromRTCPPacket(RTCPPacketInformation& rtcpPacketIn
_rtpRtcp.OnReceivedSliceLossIndication(
rtcpPacketInformation.sliPictureId);
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
{
// We need to bounce this to the default channel.
_rtpRtcp.OnReceivedEstimatedMaxBitrate(
rtcpPacketInformation.receiverEstimatedMaxBitrate);
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi)
{
// we need use a bounce it up to handle default channel

48
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -2719,7 +2719,7 @@ void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
// We received a TMMBR
const bool defaultInstance(_childModules.empty() ? false : true);
if (defaultInstance) {
ProcessDefaultModuleBandwidth(true);
ProcessDefaultModuleBandwidth();
return;
}
if (_audio) {
@ -2760,8 +2760,7 @@ void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime,
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
bool triggerOnNetworkChanged) {
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum) {
const bool defaultInstance(_childModules.empty() ? false : true);
if (!defaultInstance) {
@ -2797,22 +2796,16 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
_defaultModule->OnPacketLossStatisticsUpdate(
loss, // send in the filtered loss
roundTripTime,
lastReceivedExtendedHighSeqNum,
triggerOnNetworkChanged);
lastReceivedExtendedHighSeqNum);
}
return;
}
// No default module check if we should trigger OnNetworkChanged
// via video callback
if (triggerOnNetworkChanged)
{
_rtpReceiver.UpdateBandwidthManagement(newBitrate,
fractionLost,
roundTripTime);
}
_rtpReceiver.UpdateBandwidthManagement(newBitrate,
fractionLost,
roundTripTime);
} else {
if (!_simulcast) {
ProcessDefaultModuleBandwidth(triggerOnNetworkChanged);
ProcessDefaultModuleBandwidth();
} else {
// default and simulcast
WebRtc_UWord32 newBitrate = 0;
@ -2831,14 +2824,9 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
return;
}
_rtpSender.SetTargetSendBitrate(newBitrate);
// check if we should trigger OnNetworkChanged
// via video callback
if (triggerOnNetworkChanged)
{
_rtpReceiver.UpdateBandwidthManagement(newBitrate,
loss,
roundTripTime);
}
_rtpReceiver.UpdateBandwidthManagement(newBitrate,
loss,
roundTripTime);
// sanity
if (_sendVideoCodec.codecType == kVideoCodecUnknown) {
return;
@ -2875,8 +2863,7 @@ void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
}
}
void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth(
bool triggerOnNetworkChanged) {
void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
WebRtc_UWord32 minBitrateBps = 0xffffffff;
WebRtc_UWord32 maxBitrateBps = 0;
@ -2933,14 +2920,11 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth(
}
_bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
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);
}
// Update default module bitrate. Don't care about min max.
WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
_rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
fractionLostAvg ,
maxRoundTripTime);
}
void ModuleRtpRtcpImpl::OnRequestSendReport() {

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

@ -508,8 +508,7 @@ public:
void OnPacketLossStatisticsUpdate(
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTime,
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
bool triggerOnNetworkChanged);
const WebRtc_UWord32 lastReceivedExtendedHighSeqNum);
void OnReceivedTMMBR();
@ -562,7 +561,7 @@ protected:
RtpRtcpClock& _clock;
private:
void SendKeyFrame();
void ProcessDefaultModuleBandwidth(bool triggerOnNetworkChanged);
void ProcessDefaultModuleBandwidth();
WebRtc_Word32 _id;
const bool _audio;