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Update receive only channels with RTT.

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vie_autotest_loopback.cc will not be part of the commit, it's only to show the test.

TEST=temporary with attached loopback test.

Review URL: https://webrtc-codereview.appspot.com/390007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1678 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-02-14 12:49:59 +00:00
parent e8b1a0f188
commit d7d46887a6
3 changed files with 67 additions and 5 deletions
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20
src/modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -46,7 +46,8 @@ RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id,
_lastReceivedSRNTPsecs(0),
_lastReceivedSRNTPfrac(0),
_receivedInfoMap(),
_packetTimeOutMS(0)
_packetTimeOutMS(0),
_rtt(0)
{
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
@ -191,6 +192,23 @@ WebRtc_Word32 RTCPReceiver::RTT(const WebRtc_UWord32 remoteSSRC,
return 0;
}
WebRtc_UWord16 RTCPReceiver::RTT() const {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return 0;
}
return _rtt;
}
int RTCPReceiver::SetRTT(WebRtc_UWord16 rtt) {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return -1;
}
_rtt = rtt;
return 0;
}
void
RTCPReceiver::UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const
{

9
src/modules/rtp_rtcp/source/rtcp_receiver.h
View File

@ -69,6 +69,10 @@ public:
WebRtc_UWord16* minRTT,
WebRtc_UWord16* maxRTT) const;
WebRtc_UWord16 RTT() const;
int SetRTT(WebRtc_UWord16 rtt);
WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
void UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const;
@ -219,6 +223,11 @@ protected:
_receivedCnameMap;
WebRtc_UWord32 _packetTimeOutMS;
// Externally set RTT. This value can only be used if there are no valid
// RTT estimates.
WebRtc_UWord16 _rtt;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_

43
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -34,6 +34,8 @@ const float FracMS = 4.294967296E6f;
namespace webrtc {
const WebRtc_UWord16 kDefaultRtt = 200;
RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
const bool audio) {
return CreateRtpRtcp(id, audio, ModuleRTPUtility::GetSystemClock());
@ -378,14 +380,33 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
const bool defaultInstance(_childModules.empty() ? false : true);
if (!defaultInstance && _rtcpSender.TimeToSendRTCPReport()) {
WebRtc_UWord16 RTT = 0;
_rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
WebRtc_UWord16 max_rtt = 0;
if (_rtcpSender.Sending()) {
std::vector<RTCPReportBlock> receive_blocks;
_rtcpReceiver.StatisticsReceived(&receive_blocks);
for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
it != receive_blocks.end(); ++it) {
WebRtc_UWord16 rtt = 0;
_rtcpReceiver.RTT(it->remoteSSRC, &max_rtt, NULL, NULL, NULL);
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
}
} else {
// We're only receiving, i.e. this module doesn't have its own RTT
// estimate. Use the RTT set by a sending channel using the same default
// module.
max_rtt = _rtcpReceiver.RTT();
}
if (max_rtt == 0) {
// No valid estimate available, i.e. no sending channel using the same
// default module or no RTCP received yet.
max_rtt = kDefaultRtt;
}
if (REMB() && _rtcpSender.ValidBitrateEstimate()) {
unsigned int target_bitrate =
_rtcpSender.CalculateNewTargetBitrate(RTT);
_rtcpSender.CalculateNewTargetBitrate(max_rtt);
_rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
} else if (TMMBR()) {
_rtcpSender.CalculateNewTargetBitrate(RTT);
_rtcpSender.CalculateNewTargetBitrate(max_rtt);
}
_rtcpSender.SendRTCP(kRtcpReport);
}
@ -2758,6 +2779,20 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
// No sending modules and no bitrate estimate.
return;
}
// Update RTT to all receive only child modules, they won't have their own RTT
// estimate. Assume the receive only channels are on similar links as the
// sending channel and have approximately the same RTT.
{
CriticalSectionScoped lock(_criticalSectionModulePtrs);
for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
it != _childModules.end(); ++it) {
if (!(*it)->Sending()) {
(*it)->_rtcpReceiver.SetRTT(maxRoundTripTime);
}
}
}
_bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
// Update default module bitrate. Don't care about min max.