2f225cadde
BUG=1267 TEST=Unittest added. Review URL: https://webrtc-codereview.appspot.com/1019006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
1442 lines
50 KiB
C++
1442 lines
50 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtcp_receiver.h"
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#include <string.h> //memset
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#include <cassert> //assert
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#include "trace.h"
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#include "critical_section_wrapper.h"
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#include "rtcp_utility.h"
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#include "rtp_rtcp_impl.h"
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namespace
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{
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const float FRAC = 4.294967296E9;
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}
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namespace webrtc {
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using namespace RTCPUtility;
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using namespace RTCPHelp;
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// The number of RTCP time intervals needed to trigger a timeout.
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const int kRrTimeoutIntervals = 3;
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RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock,
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ModuleRtpRtcpImpl* owner)
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: TMMBRHelp(),
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_id(id),
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_clock(*clock),
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_method(kRtcpOff),
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_lastReceived(0),
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_rtpRtcp(*owner),
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_criticalSectionFeedbacks(
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CriticalSectionWrapper::CreateCriticalSection()),
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_cbRtcpFeedback(NULL),
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_cbRtcpBandwidthObserver(NULL),
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_cbRtcpIntraFrameObserver(NULL),
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_criticalSectionRTCPReceiver(
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CriticalSectionWrapper::CreateCriticalSection()),
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_SSRC(0),
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_remoteSSRC(0),
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_remoteSenderInfo(),
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_lastReceivedSRNTPsecs(0),
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_lastReceivedSRNTPfrac(0),
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_receivedInfoMap(),
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_packetTimeOutMS(0),
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_lastReceivedRrMs(0),
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_lastIncreasedSequenceNumberMs(0),
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_rtt(0) {
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memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
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WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
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}
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RTCPReceiver::~RTCPReceiver() {
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delete _criticalSectionRTCPReceiver;
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delete _criticalSectionFeedbacks;
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while (!_receivedReportBlockMap.empty()) {
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std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator first =
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_receivedReportBlockMap.begin();
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delete first->second;
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_receivedReportBlockMap.erase(first);
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}
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while (!_receivedInfoMap.empty()) {
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std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator first =
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_receivedInfoMap.begin();
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delete first->second;
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_receivedInfoMap.erase(first);
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}
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while (!_receivedCnameMap.empty()) {
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std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator first =
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_receivedCnameMap.begin();
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delete first->second;
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_receivedCnameMap.erase(first);
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}
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WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id,
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"%s deleted", __FUNCTION__);
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}
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void
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RTCPReceiver::ChangeUniqueId(const WebRtc_Word32 id)
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{
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_id = id;
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}
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RTCPMethod
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RTCPReceiver::Status() const
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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return _method;
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}
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WebRtc_Word32
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RTCPReceiver::SetRTCPStatus(const RTCPMethod method)
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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_method = method;
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return 0;
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}
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WebRtc_Word64
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RTCPReceiver::LastReceived()
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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return _lastReceived;
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}
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WebRtc_Word32
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RTCPReceiver::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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// new SSRC reset old reports
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memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
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_lastReceivedSRNTPsecs = 0;
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_lastReceivedSRNTPfrac = 0;
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_remoteSSRC = ssrc;
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return 0;
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}
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void RTCPReceiver::RegisterRtcpObservers(
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RtcpIntraFrameObserver* intra_frame_callback,
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RtcpBandwidthObserver* bandwidth_callback,
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RtcpFeedback* feedback_callback) {
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CriticalSectionScoped lock(_criticalSectionFeedbacks);
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_cbRtcpIntraFrameObserver = intra_frame_callback;
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_cbRtcpBandwidthObserver = bandwidth_callback;
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_cbRtcpFeedback = feedback_callback;
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}
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void RTCPReceiver::SetSSRC(const WebRtc_UWord32 ssrc) {
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WebRtc_UWord32 old_ssrc = 0;
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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old_ssrc = _SSRC;
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_SSRC = ssrc;
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}
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{
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CriticalSectionScoped lock(_criticalSectionFeedbacks);
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if (_cbRtcpIntraFrameObserver && old_ssrc != ssrc) {
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_cbRtcpIntraFrameObserver->OnLocalSsrcChanged(old_ssrc, ssrc);
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}
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}
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}
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WebRtc_Word32 RTCPReceiver::ResetRTT(const WebRtc_UWord32 remoteSSRC) {
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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RTCPReportBlockInformation* reportBlock =
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GetReportBlockInformation(remoteSSRC);
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if (reportBlock == NULL) {
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WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
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"\tfailed to GetReportBlockInformation(%u)", remoteSSRC);
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return -1;
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}
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reportBlock->RTT = 0;
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reportBlock->avgRTT = 0;
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reportBlock->minRTT = 0;
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reportBlock->maxRTT = 0;
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return 0;
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}
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WebRtc_Word32 RTCPReceiver::RTT(const WebRtc_UWord32 remoteSSRC,
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WebRtc_UWord16* RTT,
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WebRtc_UWord16* avgRTT,
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WebRtc_UWord16* minRTT,
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WebRtc_UWord16* maxRTT) const {
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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RTCPReportBlockInformation* reportBlock =
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GetReportBlockInformation(remoteSSRC);
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if (reportBlock == NULL) {
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return -1;
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}
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if (RTT) {
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*RTT = reportBlock->RTT;
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}
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if (avgRTT) {
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*avgRTT = reportBlock->avgRTT;
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}
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if (minRTT) {
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*minRTT = reportBlock->minRTT;
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}
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if (maxRTT) {
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*maxRTT = reportBlock->maxRTT;
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}
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return 0;
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}
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WebRtc_UWord16 RTCPReceiver::RTT() const {
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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if (!_receivedReportBlockMap.empty()) {
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return 0;
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}
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return _rtt;
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}
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int RTCPReceiver::SetRTT(WebRtc_UWord16 rtt) {
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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if (!_receivedReportBlockMap.empty()) {
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return -1;
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}
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_rtt = rtt;
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return 0;
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}
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WebRtc_Word32
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RTCPReceiver::NTP(WebRtc_UWord32 *ReceivedNTPsecs,
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WebRtc_UWord32 *ReceivedNTPfrac,
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WebRtc_UWord32 *RTCPArrivalTimeSecs,
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WebRtc_UWord32 *RTCPArrivalTimeFrac,
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WebRtc_UWord32 *rtcp_timestamp) const
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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if(ReceivedNTPsecs)
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{
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*ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport
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}
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if(ReceivedNTPfrac)
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{
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*ReceivedNTPfrac = _remoteSenderInfo.NTPfraction;
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}
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if(RTCPArrivalTimeFrac)
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{
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*RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block
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}
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if(RTCPArrivalTimeSecs)
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{
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*RTCPArrivalTimeSecs = _lastReceivedSRNTPsecs;
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}
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if (rtcp_timestamp) {
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*rtcp_timestamp = _remoteSenderInfo.RTPtimeStamp;
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}
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return 0;
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}
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WebRtc_Word32
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RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const
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{
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if(senderInfo == NULL)
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{
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WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
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return -1;
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}
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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if(_lastReceivedSRNTPsecs == 0)
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{
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WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__);
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return -1;
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}
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memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
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return 0;
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}
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// statistics
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// we can get multiple receive reports when we receive the report from a CE
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WebRtc_Word32 RTCPReceiver::StatisticsReceived(
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std::vector<RTCPReportBlock>* receiveBlocks) const {
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assert(receiveBlocks);
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
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_receivedReportBlockMap.begin();
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while (it != _receivedReportBlockMap.end()) {
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receiveBlocks->push_back(it->second->remoteReceiveBlock);
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it++;
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}
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return 0;
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}
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WebRtc_Word32
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RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
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RTCPUtility::RTCPParserV2* rtcpParser)
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{
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CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
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_lastReceived = _clock.GetTimeInMS();
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RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
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while (pktType != RTCPUtility::kRtcpNotValidCode)
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{
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// Each "case" is responsible for iterate the parser to the
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// next top level packet.
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switch (pktType)
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{
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case RTCPUtility::kRtcpSrCode:
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case RTCPUtility::kRtcpRrCode:
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HandleSenderReceiverReport(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpSdesCode:
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HandleSDES(*rtcpParser);
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break;
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case RTCPUtility::kRtcpXrVoipMetricCode:
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HandleXRVOIPMetric(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpByeCode:
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HandleBYE(*rtcpParser);
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break;
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case RTCPUtility::kRtcpRtpfbNackCode:
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HandleNACK(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpRtpfbTmmbrCode:
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HandleTMMBR(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpRtpfbTmmbnCode:
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HandleTMMBN(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpRtpfbSrReqCode:
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HandleSR_REQ(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpPsfbPliCode:
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HandlePLI(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpPsfbSliCode:
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HandleSLI(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpPsfbRpsiCode:
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HandleRPSI(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpExtendedIjCode:
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HandleIJ(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpPsfbFirCode:
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HandleFIR(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpPsfbAppCode:
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HandlePsfbApp(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpAppCode:
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// generic application messages
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HandleAPP(*rtcpParser, rtcpPacketInformation);
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break;
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case RTCPUtility::kRtcpAppItemCode:
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// generic application messages
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HandleAPPItem(*rtcpParser, rtcpPacketInformation);
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break;
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default:
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rtcpParser->Iterate();
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break;
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}
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pktType = rtcpParser->PacketType();
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}
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return 0;
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}
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// no need for critsect we have _criticalSectionRTCPReceiver
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void
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RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
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RTCPPacketInformation& rtcpPacketInformation)
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{
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RTCPUtility::RTCPPacketTypes rtcpPacketType = rtcpParser.PacketType();
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const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
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assert((rtcpPacketType == RTCPUtility::kRtcpRrCode) || (rtcpPacketType == RTCPUtility::kRtcpSrCode));
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// SR.SenderSSRC
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// The synchronization source identifier for the originator of this SR packet
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// rtcpPacket.RR.SenderSSRC
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// The source of the packet sender, same as of SR? or is this a CE?
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const WebRtc_UWord32 remoteSSRC = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.SenderSSRC:rtcpPacket.SR.SenderSSRC;
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const WebRtc_UWord8 numberOfReportBlocks = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.NumberOfReportBlocks:rtcpPacket.SR.NumberOfReportBlocks;
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rtcpPacketInformation.remoteSSRC = remoteSSRC;
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RTCPReceiveInformation* ptrReceiveInfo = CreateReceiveInformation(remoteSSRC);
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if (!ptrReceiveInfo)
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{
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rtcpParser.Iterate();
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return;
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}
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if (rtcpPacketType == RTCPUtility::kRtcpSrCode)
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{
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WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
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"Received SR(%d). SSRC:0x%x, from SSRC:0x%x, to us %d.", _id, _SSRC, remoteSSRC, (_remoteSSRC == remoteSSRC)?1:0);
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if (_remoteSSRC == remoteSSRC) // have I received RTP packets from this party
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{
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// only signal that we have received a SR when we accept one
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rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSr;
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rtcpPacketInformation.ntp_secs = rtcpPacket.SR.NTPMostSignificant;
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rtcpPacketInformation.ntp_frac = rtcpPacket.SR.NTPLeastSignificant;
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rtcpPacketInformation.rtp_timestamp = rtcpPacket.SR.RTPTimestamp;
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// We will only store the send report from one source, but
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// we will store all the receive block
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// Save the NTP time of this report
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_remoteSenderInfo.NTPseconds = rtcpPacket.SR.NTPMostSignificant;
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_remoteSenderInfo.NTPfraction = rtcpPacket.SR.NTPLeastSignificant;
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_remoteSenderInfo.RTPtimeStamp = rtcpPacket.SR.RTPTimestamp;
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_remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount;
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_remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount;
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_clock.CurrentNTP(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
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}
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else
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{
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rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
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}
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} else
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{
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WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
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"Received RR(%d). SSRC:0x%x, from SSRC:0x%x", _id, _SSRC, remoteSSRC);
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rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
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}
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UpdateReceiveInformation(*ptrReceiveInfo);
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rtcpPacketType = rtcpParser.Iterate();
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while (rtcpPacketType == RTCPUtility::kRtcpReportBlockItemCode)
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{
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HandleReportBlock(rtcpPacket, rtcpPacketInformation, remoteSSRC, numberOfReportBlocks);
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rtcpPacketType = rtcpParser.Iterate();
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}
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}
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// no need for critsect we have _criticalSectionRTCPReceiver
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void
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RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
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RTCPPacketInformation& rtcpPacketInformation,
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const WebRtc_UWord32 remoteSSRC,
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const WebRtc_UWord8 numberOfReportBlocks) {
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// This will be called once per report block in the RTCP packet.
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// We filter out all report blocks that are not for us.
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// Each packet has max 31 RR blocks.
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//
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// We can calc RTT if we send a send report and get a report block back.
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// |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to
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// which the information in this reception report block pertains.
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// Filter out all report blocks that are not for us.
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if (rtcpPacket.ReportBlockItem.SSRC != _SSRC) {
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// This block is not for us ignore it.
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return;
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}
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// To avoid problem with acquiring _criticalSectionRTCPSender while holding
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// _criticalSectionRTCPReceiver.
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_criticalSectionRTCPReceiver->Leave();
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WebRtc_UWord32 sendTimeMS =
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_rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR);
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_criticalSectionRTCPReceiver->Enter();
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RTCPReportBlockInformation* reportBlock =
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CreateReportBlockInformation(remoteSSRC);
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if (reportBlock == NULL) {
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WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
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"\tfailed to CreateReportBlockInformation(%u)", remoteSSRC);
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return;
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}
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_lastReceivedRrMs = _clock.GetTimeInMS();
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const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem;
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reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC;
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reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
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reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost;
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reportBlock->remoteReceiveBlock.cumulativeLost =
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rb.CumulativeNumOfPacketsLost;
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if (rb.ExtendedHighestSequenceNumber >
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reportBlock->remoteReceiveBlock.extendedHighSeqNum) {
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// We have successfully delivered new RTP packets to the remote side after
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// the last RR was sent from the remote side.
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_lastIncreasedSequenceNumberMs = _lastReceivedRrMs;
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}
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reportBlock->remoteReceiveBlock.extendedHighSeqNum =
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rb.ExtendedHighestSequenceNumber;
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reportBlock->remoteReceiveBlock.jitter = rb.Jitter;
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reportBlock->remoteReceiveBlock.delaySinceLastSR = rb.DelayLastSR;
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reportBlock->remoteReceiveBlock.lastSR = rb.LastSR;
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if (rtcpPacket.ReportBlockItem.Jitter > reportBlock->remoteMaxJitter) {
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reportBlock->remoteMaxJitter = rtcpPacket.ReportBlockItem.Jitter;
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|
}
|
|
|
|
WebRtc_UWord32 delaySinceLastSendReport =
|
|
rtcpPacket.ReportBlockItem.DelayLastSR;
|
|
|
|
// local NTP time when we received this
|
|
WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
|
|
WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
|
|
|
|
_clock.CurrentNTP(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
|
|
|
|
// time when we received this in MS
|
|
WebRtc_UWord32 receiveTimeMS = ModuleRTPUtility::ConvertNTPTimeToMS(
|
|
lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
|
|
|
|
// Estimate RTT
|
|
WebRtc_UWord32 d = (delaySinceLastSendReport & 0x0000ffff) * 1000;
|
|
d /= 65536;
|
|
d += ((delaySinceLastSendReport & 0xffff0000) >> 16) * 1000;
|
|
|
|
WebRtc_Word32 RTT = 0;
|
|
|
|
if (sendTimeMS > 0) {
|
|
RTT = receiveTimeMS - d - sendTimeMS;
|
|
if (RTT <= 0) {
|
|
RTT = 1;
|
|
}
|
|
if (RTT > reportBlock->maxRTT) {
|
|
// store max RTT
|
|
reportBlock->maxRTT = (WebRtc_UWord16) RTT;
|
|
}
|
|
if (reportBlock->minRTT == 0) {
|
|
// first RTT
|
|
reportBlock->minRTT = (WebRtc_UWord16) RTT;
|
|
} else if (RTT < reportBlock->minRTT) {
|
|
// Store min RTT
|
|
reportBlock->minRTT = (WebRtc_UWord16) RTT;
|
|
}
|
|
// store last RTT
|
|
reportBlock->RTT = (WebRtc_UWord16) RTT;
|
|
|
|
// store average RTT
|
|
if (reportBlock->numAverageCalcs != 0) {
|
|
float ac = static_cast<float> (reportBlock->numAverageCalcs);
|
|
float newAverage = ((ac / (ac + 1)) * reportBlock->avgRTT)
|
|
+ ((1 / (ac + 1)) * RTT);
|
|
reportBlock->avgRTT = static_cast<int> (newAverage + 0.5f);
|
|
} else {
|
|
// first RTT
|
|
reportBlock->avgRTT = (WebRtc_UWord16) RTT;
|
|
}
|
|
reportBlock->numAverageCalcs++;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
|
|
" -> Received report block(%d), from SSRC:0x%x, RTT:%d, loss:%d",
|
|
_id, remoteSSRC, RTT, rtcpPacket.ReportBlockItem.FractionLost);
|
|
|
|
// rtcpPacketInformation
|
|
rtcpPacketInformation.AddReportInfo(
|
|
reportBlock->remoteReceiveBlock.fractionLost, (WebRtc_UWord16) RTT,
|
|
reportBlock->remoteReceiveBlock.extendedHighSeqNum,
|
|
reportBlock->remoteReceiveBlock.jitter);
|
|
}
|
|
|
|
RTCPReportBlockInformation*
|
|
RTCPReceiver::CreateReportBlockInformation(WebRtc_UWord32 remoteSSRC) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator it =
|
|
_receivedReportBlockMap.find(remoteSSRC);
|
|
|
|
RTCPReportBlockInformation* ptrReportBlockInfo = NULL;
|
|
if (it != _receivedReportBlockMap.end()) {
|
|
ptrReportBlockInfo = it->second;
|
|
} else {
|
|
ptrReportBlockInfo = new RTCPReportBlockInformation;
|
|
_receivedReportBlockMap[remoteSSRC] = ptrReportBlockInfo;
|
|
}
|
|
return ptrReportBlockInfo;
|
|
}
|
|
|
|
RTCPReportBlockInformation*
|
|
RTCPReceiver::GetReportBlockInformation(WebRtc_UWord32 remoteSSRC) const {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
|
|
_receivedReportBlockMap.find(remoteSSRC);
|
|
|
|
if (it == _receivedReportBlockMap.end()) {
|
|
return NULL;
|
|
}
|
|
return it->second;
|
|
}
|
|
|
|
RTCPCnameInformation*
|
|
RTCPReceiver::CreateCnameInformation(WebRtc_UWord32 remoteSSRC) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
|
|
_receivedCnameMap.find(remoteSSRC);
|
|
|
|
if (it != _receivedCnameMap.end()) {
|
|
return it->second;
|
|
}
|
|
RTCPCnameInformation* cnameInfo = new RTCPCnameInformation;
|
|
memset(cnameInfo->name, 0, RTCP_CNAME_SIZE);
|
|
_receivedCnameMap[remoteSSRC] = cnameInfo;
|
|
return cnameInfo;
|
|
}
|
|
|
|
RTCPCnameInformation*
|
|
RTCPReceiver::GetCnameInformation(WebRtc_UWord32 remoteSSRC) const {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPCnameInformation*>::const_iterator it =
|
|
_receivedCnameMap.find(remoteSSRC);
|
|
|
|
if (it == _receivedCnameMap.end()) {
|
|
return NULL;
|
|
}
|
|
return it->second;
|
|
}
|
|
|
|
RTCPReceiveInformation*
|
|
RTCPReceiver::CreateReceiveInformation(WebRtc_UWord32 remoteSSRC) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
|
|
_receivedInfoMap.find(remoteSSRC);
|
|
|
|
if (it != _receivedInfoMap.end()) {
|
|
return it->second;
|
|
}
|
|
RTCPReceiveInformation* receiveInfo = new RTCPReceiveInformation;
|
|
_receivedInfoMap[remoteSSRC] = receiveInfo;
|
|
return receiveInfo;
|
|
}
|
|
|
|
RTCPReceiveInformation*
|
|
RTCPReceiver::GetReceiveInformation(WebRtc_UWord32 remoteSSRC) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
|
|
_receivedInfoMap.find(remoteSSRC);
|
|
if (it == _receivedInfoMap.end()) {
|
|
return NULL;
|
|
}
|
|
return it->second;
|
|
}
|
|
|
|
void RTCPReceiver::UpdateReceiveInformation(
|
|
RTCPReceiveInformation& receiveInformation) {
|
|
// Update that this remote is alive
|
|
receiveInformation.lastTimeReceived = _clock.GetTimeInMS();
|
|
}
|
|
|
|
bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
if (_lastReceivedRrMs == 0)
|
|
return false;
|
|
|
|
int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
|
|
if (_clock.GetTimeInMS() > _lastReceivedRrMs + time_out_ms) {
|
|
// Reset the timer to only trigger one log.
|
|
_lastReceivedRrMs = 0;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
if (_lastIncreasedSequenceNumberMs == 0)
|
|
return false;
|
|
|
|
int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
|
|
if (_clock.GetTimeInMS() > _lastIncreasedSequenceNumberMs + time_out_ms) {
|
|
// Reset the timer to only trigger one log.
|
|
_lastIncreasedSequenceNumberMs = 0;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
bool updateBoundingSet = false;
|
|
WebRtc_Word64 timeNow = _clock.GetTimeInMS();
|
|
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
|
|
_receivedInfoMap.begin();
|
|
|
|
while (receiveInfoIt != _receivedInfoMap.end()) {
|
|
RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
|
|
if (receiveInfo == NULL) {
|
|
return updateBoundingSet;
|
|
}
|
|
// time since last received rtcp packet
|
|
// when we dont have a lastTimeReceived and the object is marked
|
|
// readyForDelete it's removed from the map
|
|
if (receiveInfo->lastTimeReceived) {
|
|
/// use audio define since we don't know what interval the remote peer is
|
|
// using
|
|
if ((timeNow - receiveInfo->lastTimeReceived) >
|
|
5 * RTCP_INTERVAL_AUDIO_MS) {
|
|
// no rtcp packet for the last five regular intervals, reset limitations
|
|
receiveInfo->TmmbrSet.clearSet();
|
|
// prevent that we call this over and over again
|
|
receiveInfo->lastTimeReceived = 0;
|
|
// send new TMMBN to all channels using the default codec
|
|
updateBoundingSet = true;
|
|
}
|
|
receiveInfoIt++;
|
|
} else if (receiveInfo->readyForDelete) {
|
|
// store our current receiveInfoItem
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator
|
|
receiveInfoItemToBeErased = receiveInfoIt;
|
|
receiveInfoIt++;
|
|
delete receiveInfoItemToBeErased->second;
|
|
_receivedInfoMap.erase(receiveInfoItemToBeErased);
|
|
} else {
|
|
receiveInfoIt++;
|
|
}
|
|
}
|
|
return updateBoundingSet;
|
|
}
|
|
|
|
WebRtc_Word32 RTCPReceiver::BoundingSet(bool &tmmbrOwner,
|
|
TMMBRSet* boundingSetRec) {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
|
|
_receivedInfoMap.find(_remoteSSRC);
|
|
|
|
if (receiveInfoIt == _receivedInfoMap.end()) {
|
|
return -1;
|
|
}
|
|
RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
|
|
if (receiveInfo == NULL) {
|
|
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
|
|
"%s failed to get RTCPReceiveInformation",
|
|
__FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (receiveInfo->TmmbnBoundingSet.lengthOfSet() > 0) {
|
|
boundingSetRec->VerifyAndAllocateSet(
|
|
receiveInfo->TmmbnBoundingSet.lengthOfSet() + 1);
|
|
for(WebRtc_UWord32 i=0; i< receiveInfo->TmmbnBoundingSet.lengthOfSet();
|
|
i++) {
|
|
if(receiveInfo->TmmbnBoundingSet.Ssrc(i) == _SSRC) {
|
|
// owner of bounding set
|
|
tmmbrOwner = true;
|
|
}
|
|
boundingSetRec->SetEntry(i,
|
|
receiveInfo->TmmbnBoundingSet.Tmmbr(i),
|
|
receiveInfo->TmmbnBoundingSet.PacketOH(i),
|
|
receiveInfo->TmmbnBoundingSet.Ssrc(i));
|
|
}
|
|
}
|
|
return receiveInfo->TmmbnBoundingSet.lengthOfSet();
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser)
|
|
{
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpSdesChunkCode)
|
|
{
|
|
HandleSDESChunk(rtcpParser);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
RTCPCnameInformation* cnameInfo =
|
|
CreateCnameInformation(rtcpPacket.CName.SenderSSRC);
|
|
assert(cnameInfo);
|
|
|
|
cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
|
|
strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
if (_SSRC != rtcpPacket.NACK.MediaSSRC)
|
|
{
|
|
// Not to us.
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
|
|
rtcpPacketInformation.ResetNACKPacketIdArray();
|
|
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpRtpfbNackItemCode)
|
|
{
|
|
HandleNACKItem(rtcpPacket, rtcpPacketInformation);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID);
|
|
|
|
WebRtc_UWord16 bitMask = rtcpPacket.NACKItem.BitMask;
|
|
if(bitMask)
|
|
{
|
|
for(int i=1; i <= 16; ++i)
|
|
{
|
|
if(bitMask & 0x01)
|
|
{
|
|
rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID + i);
|
|
}
|
|
bitMask = bitMask >>1;
|
|
}
|
|
}
|
|
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpNack;
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser) {
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
// clear our lists
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator
|
|
reportBlockInfoIt = _receivedReportBlockMap.find(
|
|
rtcpPacket.BYE.SenderSSRC);
|
|
|
|
if (reportBlockInfoIt != _receivedReportBlockMap.end()) {
|
|
delete reportBlockInfoIt->second;
|
|
_receivedReportBlockMap.erase(reportBlockInfoIt);
|
|
}
|
|
// we can't delete it due to TMMBR
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
|
|
_receivedInfoMap.find(rtcpPacket.BYE.SenderSSRC);
|
|
|
|
if (receiveInfoIt != _receivedInfoMap.end()) {
|
|
receiveInfoIt->second->readyForDelete = true;
|
|
}
|
|
|
|
std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator cnameInfoIt =
|
|
_receivedCnameMap.find(rtcpPacket.BYE.SenderSSRC);
|
|
|
|
if (cnameInfoIt != _receivedCnameMap.end()) {
|
|
delete cnameInfoIt->second;
|
|
_receivedCnameMap.erase(cnameInfoIt);
|
|
}
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
if(rtcpPacket.XRVOIPMetricItem.SSRC == _SSRC)
|
|
{
|
|
// Store VoIP metrics block if it's about me
|
|
// from OriginatorSSRC do we filter it?
|
|
// rtcpPacket.XR.OriginatorSSRC;
|
|
|
|
RTCPVoIPMetric receivedVoIPMetrics;
|
|
receivedVoIPMetrics.burstDensity = rtcpPacket.XRVOIPMetricItem.burstDensity;
|
|
receivedVoIPMetrics.burstDuration = rtcpPacket.XRVOIPMetricItem.burstDuration;
|
|
receivedVoIPMetrics.discardRate = rtcpPacket.XRVOIPMetricItem.discardRate;
|
|
receivedVoIPMetrics.endSystemDelay = rtcpPacket.XRVOIPMetricItem.endSystemDelay;
|
|
receivedVoIPMetrics.extRfactor = rtcpPacket.XRVOIPMetricItem.extRfactor;
|
|
receivedVoIPMetrics.gapDensity = rtcpPacket.XRVOIPMetricItem.gapDensity;
|
|
receivedVoIPMetrics.gapDuration = rtcpPacket.XRVOIPMetricItem.gapDuration;
|
|
receivedVoIPMetrics.Gmin = rtcpPacket.XRVOIPMetricItem.Gmin;
|
|
receivedVoIPMetrics.JBabsMax = rtcpPacket.XRVOIPMetricItem.JBabsMax;
|
|
receivedVoIPMetrics.JBmax = rtcpPacket.XRVOIPMetricItem.JBmax;
|
|
receivedVoIPMetrics.JBnominal = rtcpPacket.XRVOIPMetricItem.JBnominal;
|
|
receivedVoIPMetrics.lossRate = rtcpPacket.XRVOIPMetricItem.lossRate;
|
|
receivedVoIPMetrics.MOSCQ = rtcpPacket.XRVOIPMetricItem.MOSCQ;
|
|
receivedVoIPMetrics.MOSLQ = rtcpPacket.XRVOIPMetricItem.MOSLQ;
|
|
receivedVoIPMetrics.noiseLevel = rtcpPacket.XRVOIPMetricItem.noiseLevel;
|
|
receivedVoIPMetrics.RERL = rtcpPacket.XRVOIPMetricItem.RERL;
|
|
receivedVoIPMetrics.Rfactor = rtcpPacket.XRVOIPMetricItem.Rfactor;
|
|
receivedVoIPMetrics.roundTripDelay = rtcpPacket.XRVOIPMetricItem.roundTripDelay;
|
|
receivedVoIPMetrics.RXconfig = rtcpPacket.XRVOIPMetricItem.RXconfig;
|
|
receivedVoIPMetrics.signalLevel = rtcpPacket.XRVOIPMetricItem.signalLevel;
|
|
|
|
rtcpPacketInformation.AddVoIPMetric(&receivedVoIPMetrics);
|
|
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrVoipMetric; // received signal
|
|
}
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
if (_SSRC == rtcpPacket.PLI.MediaSSRC) {
|
|
// Received a signal that we need to send a new key frame.
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli;
|
|
}
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
WebRtc_UWord32 senderSSRC = rtcpPacket.TMMBR.SenderSSRC;
|
|
RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(senderSSRC);
|
|
if (ptrReceiveInfo == NULL)
|
|
{
|
|
// This remote SSRC must be saved before.
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
if(rtcpPacket.TMMBR.MediaSSRC)
|
|
{
|
|
// rtcpPacket.TMMBR.MediaSSRC SHOULD be 0 if same as SenderSSRC
|
|
// in relay mode this is a valid number
|
|
senderSSRC = rtcpPacket.TMMBR.MediaSSRC;
|
|
}
|
|
|
|
// Use packet length to calc max number of TMMBR blocks
|
|
// each TMMBR block is 8 bytes
|
|
ptrdiff_t maxNumOfTMMBRBlocks = rtcpParser.LengthLeft() / 8;
|
|
|
|
// sanity
|
|
if(maxNumOfTMMBRBlocks > 200) // we can't have more than what's in one packet
|
|
{
|
|
assert(false);
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
ptrReceiveInfo->VerifyAndAllocateTMMBRSet((WebRtc_UWord32)maxNumOfTMMBRBlocks);
|
|
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpRtpfbTmmbrItemCode)
|
|
{
|
|
HandleTMMBRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation, senderSSRC);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
|
|
const RTCPUtility::RTCPPacket& rtcpPacket,
|
|
RTCPPacketInformation& rtcpPacketInformation,
|
|
const WebRtc_UWord32 senderSSRC)
|
|
{
|
|
if (_SSRC == rtcpPacket.TMMBRItem.SSRC &&
|
|
rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0)
|
|
{
|
|
receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem,
|
|
_clock.GetTimeInMS());
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr;
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.TMMBN.SenderSSRC);
|
|
if (ptrReceiveInfo == NULL)
|
|
{
|
|
// This remote SSRC must be saved before.
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbn;
|
|
// Use packet length to calc max number of TMMBN blocks
|
|
// each TMMBN block is 8 bytes
|
|
ptrdiff_t maxNumOfTMMBNBlocks = rtcpParser.LengthLeft() / 8;
|
|
|
|
// sanity
|
|
if(maxNumOfTMMBNBlocks > 200) // we cant have more than what's in one packet
|
|
{
|
|
assert(false);
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
|
|
ptrReceiveInfo->VerifyAndAllocateBoundingSet((WebRtc_UWord32)maxNumOfTMMBNBlocks);
|
|
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode)
|
|
{
|
|
HandleTMMBNItem(*ptrReceiveInfo, rtcpPacket);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSrReq;
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo,
|
|
const RTCPUtility::RTCPPacket& rtcpPacket)
|
|
{
|
|
receiveInfo.TmmbnBoundingSet.AddEntry(
|
|
rtcpPacket.TMMBNItem.MaxTotalMediaBitRate,
|
|
rtcpPacket.TMMBNItem.MeasuredOverhead,
|
|
rtcpPacket.TMMBNItem.SSRC);
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpPsfbSliItemCode)
|
|
{
|
|
HandleSLIItem(rtcpPacket, rtcpPacketInformation);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
// in theory there could be multiple slices lost
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSli; // received signal that we need to refresh a slice
|
|
rtcpPacketInformation.sliPictureId = rtcpPacket.SLIItem.PictureId;
|
|
}
|
|
|
|
void
|
|
RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
if(pktType == RTCPUtility::kRtcpPsfbRpsiCode)
|
|
{
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRpsi; // received signal that we have a confirmed reference picture
|
|
if(rtcpPacket.RPSI.NumberOfValidBits%8 != 0)
|
|
{
|
|
// to us unknown
|
|
// continue
|
|
rtcpParser.Iterate();
|
|
return;
|
|
}
|
|
rtcpPacketInformation.rpsiPictureId = 0;
|
|
|
|
// convert NativeBitString to rpsiPictureId
|
|
WebRtc_UWord8 numberOfBytes = rtcpPacket.RPSI.NumberOfValidBits /8;
|
|
for(WebRtc_UWord8 n = 0; n < (numberOfBytes-1); n++)
|
|
{
|
|
rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[n] & 0x7f);
|
|
rtcpPacketInformation.rpsiPictureId <<= 7; // prepare next
|
|
}
|
|
rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[numberOfBytes-1] & 0x7f);
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
if (pktType == RTCPUtility::kRtcpPsfbRembCode) {
|
|
pktType = rtcpParser.Iterate();
|
|
if (pktType == RTCPUtility::kRtcpPsfbRembItemCode) {
|
|
HandleREMBItem(rtcpParser, rtcpPacketInformation);
|
|
rtcpParser.Iterate();
|
|
}
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void
|
|
RTCPReceiver::HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpExtendedIjItemCode)
|
|
{
|
|
HandleIJItem(rtcpPacket, rtcpPacketInformation);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
void
|
|
RTCPReceiver::HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
|
|
rtcpPacketInformation.interArrivalJitter =
|
|
rtcpPacket.ExtendedJitterReportItem.Jitter;
|
|
}
|
|
|
|
void RTCPReceiver::HandleREMBItem(
|
|
RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb;
|
|
rtcpPacketInformation.receiverEstimatedMaxBitrate =
|
|
rtcpPacket.REMBItem.BitRate;
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
RTCPReceiveInformation* ptrReceiveInfo =
|
|
GetReceiveInformation(rtcpPacket.FIR.SenderSSRC);
|
|
|
|
RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
|
|
while (pktType == RTCPUtility::kRtcpPsfbFirItemCode) {
|
|
HandleFIRItem(ptrReceiveInfo, rtcpPacket, rtcpPacketInformation);
|
|
pktType = rtcpParser.Iterate();
|
|
}
|
|
}
|
|
|
|
// no need for critsect we have _criticalSectionRTCPReceiver
|
|
void RTCPReceiver::HandleFIRItem(RTCPReceiveInformation* receiveInfo,
|
|
const RTCPUtility::RTCPPacket& rtcpPacket,
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
// Is it our sender that is requested to generate a new keyframe
|
|
if (_SSRC != rtcpPacket.FIRItem.SSRC) {
|
|
return;
|
|
}
|
|
// rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it
|
|
// we don't know who this originate from
|
|
if (receiveInfo) {
|
|
// check if we have reported this FIRSequenceNumber before
|
|
if (rtcpPacket.FIRItem.CommandSequenceNumber !=
|
|
receiveInfo->lastFIRSequenceNumber) {
|
|
WebRtc_Word64 now = _clock.GetTimeInMS();
|
|
// sanity; don't go crazy with the callbacks
|
|
if ((now - receiveInfo->lastFIRRequest) > RTCP_MIN_FRAME_LENGTH_MS) {
|
|
receiveInfo->lastFIRRequest = now;
|
|
receiveInfo->lastFIRSequenceNumber =
|
|
rtcpPacket.FIRItem.CommandSequenceNumber;
|
|
// received signal that we need to send a new key frame
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir;
|
|
}
|
|
}
|
|
} else {
|
|
// received signal that we need to send a new key frame
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir;
|
|
}
|
|
}
|
|
|
|
void
|
|
RTCPReceiver::HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpApp;
|
|
rtcpPacketInformation.applicationSubType = rtcpPacket.APP.SubType;
|
|
rtcpPacketInformation.applicationName = rtcpPacket.APP.Name;
|
|
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
void
|
|
RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
|
|
RTCPPacketInformation& rtcpPacketInformation)
|
|
{
|
|
const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
|
|
|
|
rtcpPacketInformation.AddApplicationData(rtcpPacket.APP.Data, rtcpPacket.APP.Size);
|
|
|
|
rtcpParser.Iterate();
|
|
}
|
|
|
|
WebRtc_Word32 RTCPReceiver::UpdateTMMBR() {
|
|
WebRtc_Word32 numBoundingSet = 0;
|
|
WebRtc_UWord32 bitrate = 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(&bitrate)) {
|
|
// we have a new bandwidth estimate on this channel
|
|
CriticalSectionScoped lock(_criticalSectionFeedbacks);
|
|
if (_cbRtcpBandwidthObserver) {
|
|
_cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000);
|
|
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
|
|
"Set TMMBR request:%d kbps", bitrate);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
// Holding no Critical section
|
|
void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
|
|
RTCPPacketInformation& rtcpPacketInformation) {
|
|
// 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.
|
|
UpdateTMMBR();
|
|
}
|
|
unsigned int local_ssrc = 0;
|
|
{
|
|
// We don't want to hold this critsect when triggering the callbacks below.
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
local_ssrc = _SSRC;
|
|
}
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq) {
|
|
_rtpRtcp.OnRequestSendReport();
|
|
}
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
|
|
if (rtcpPacketInformation.nackSequenceNumbersLength > 0) {
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
|
|
"SIG [RTCP] Incoming NACK length:%d",
|
|
rtcpPacketInformation.nackSequenceNumbersLength);
|
|
_rtpRtcp.OnReceivedNACK(
|
|
rtcpPacketInformation.nackSequenceNumbersLength,
|
|
rtcpPacketInformation.nackSequenceNumbers);
|
|
}
|
|
}
|
|
{
|
|
CriticalSectionScoped lock(_criticalSectionFeedbacks);
|
|
|
|
// We need feedback that we have received a report block(s) so that we
|
|
// can generate a new packet in a conference relay scenario, one received
|
|
// report can generate several RTCP packets, based on number relayed/mixed
|
|
// a send report block should go out to all receivers.
|
|
if (_cbRtcpIntraFrameObserver) {
|
|
if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
|
|
(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) {
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) {
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
|
|
"SIG [RTCP] Incoming PLI from SSRC:0x%x",
|
|
rtcpPacketInformation.remoteSSRC);
|
|
} else {
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
|
|
"SIG [RTCP] Incoming FIR from SSRC:0x%x",
|
|
rtcpPacketInformation.remoteSSRC);
|
|
}
|
|
_cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc);
|
|
}
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSli) {
|
|
_cbRtcpIntraFrameObserver->OnReceivedSLI(
|
|
local_ssrc, rtcpPacketInformation.sliPictureId);
|
|
}
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi) {
|
|
_cbRtcpIntraFrameObserver->OnReceivedRPSI(
|
|
local_ssrc, rtcpPacketInformation.rpsiPictureId);
|
|
}
|
|
}
|
|
if (_cbRtcpBandwidthObserver) {
|
|
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) {
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
|
|
"SIG [RTCP] Incoming REMB:%d",
|
|
rtcpPacketInformation.receiverEstimatedMaxBitrate);
|
|
_cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(
|
|
rtcpPacketInformation.receiverEstimatedMaxBitrate);
|
|
}
|
|
if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
|
|
rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr) &&
|
|
rtcpPacketInformation.reportBlock) {
|
|
WebRtc_Word64 now = _clock.GetTimeInMS();
|
|
_cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport(
|
|
rtcpPacketInformation.remoteSSRC,
|
|
rtcpPacketInformation.fractionLost,
|
|
rtcpPacketInformation.roundTripTime,
|
|
rtcpPacketInformation.lastReceivedExtendedHighSeqNum,
|
|
now);
|
|
}
|
|
}
|
|
if(_cbRtcpFeedback) {
|
|
if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) {
|
|
_cbRtcpFeedback->OnSendReportReceived(_id,
|
|
rtcpPacketInformation.remoteSSRC,
|
|
rtcpPacketInformation.ntp_secs,
|
|
rtcpPacketInformation.ntp_frac,
|
|
rtcpPacketInformation.rtp_timestamp);
|
|
} else {
|
|
_cbRtcpFeedback->OnReceiveReportReceived(_id,
|
|
rtcpPacketInformation.remoteSSRC);
|
|
}
|
|
if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric) {
|
|
_cbRtcpFeedback->OnXRVoIPMetricReceived(_id,
|
|
rtcpPacketInformation.VoIPMetric);
|
|
}
|
|
if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp) {
|
|
_cbRtcpFeedback->OnApplicationDataReceived(_id,
|
|
rtcpPacketInformation.applicationSubType,
|
|
rtcpPacketInformation.applicationName,
|
|
rtcpPacketInformation.applicationLength,
|
|
rtcpPacketInformation.applicationData);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
|
|
char cName[RTCP_CNAME_SIZE]) const {
|
|
assert(cName);
|
|
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
|
|
if (cnameInfo == NULL) {
|
|
return -1;
|
|
}
|
|
cName[RTCP_CNAME_SIZE - 1] = 0;
|
|
strncpy(cName, cnameInfo->name, RTCP_CNAME_SIZE - 1);
|
|
return 0;
|
|
}
|
|
|
|
// no callbacks allowed inside this function
|
|
WebRtc_Word32 RTCPReceiver::TMMBRReceived(const WebRtc_UWord32 size,
|
|
const WebRtc_UWord32 accNumCandidates,
|
|
TMMBRSet* candidateSet) const {
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
|
|
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::const_iterator
|
|
receiveInfoIt = _receivedInfoMap.begin();
|
|
if (receiveInfoIt == _receivedInfoMap.end()) {
|
|
return -1;
|
|
}
|
|
WebRtc_UWord32 num = accNumCandidates;
|
|
if (candidateSet) {
|
|
while( num < size && receiveInfoIt != _receivedInfoMap.end()) {
|
|
RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
|
|
if (receiveInfo == NULL) {
|
|
return 0;
|
|
}
|
|
for (WebRtc_UWord32 i = 0;
|
|
(num < size) && (i < receiveInfo->TmmbrSet.lengthOfSet()); i++) {
|
|
if (receiveInfo->GetTMMBRSet(i, num, candidateSet,
|
|
_clock.GetTimeInMS()) == 0) {
|
|
num++;
|
|
}
|
|
}
|
|
receiveInfoIt++;
|
|
}
|
|
} else {
|
|
while (receiveInfoIt != _receivedInfoMap.end()) {
|
|
RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
|
|
if(receiveInfo == NULL) {
|
|
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
|
|
"%s failed to get RTCPReceiveInformation",
|
|
__FUNCTION__);
|
|
return -1;
|
|
}
|
|
num += receiveInfo->TmmbrSet.lengthOfSet();
|
|
receiveInfoIt++;
|
|
}
|
|
}
|
|
return num;
|
|
}
|
|
|
|
WebRtc_Word32
|
|
RTCPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
|
|
{
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
_packetTimeOutMS = timeoutMS;
|
|
return 0;
|
|
}
|
|
|
|
void RTCPReceiver::PacketTimeout()
|
|
{
|
|
if(_packetTimeOutMS == 0)
|
|
{
|
|
// not configured
|
|
return;
|
|
}
|
|
|
|
bool packetTimeOut = false;
|
|
{
|
|
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
|
|
if(_lastReceived == 0)
|
|
{
|
|
// not active
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word64 now = _clock.GetTimeInMS();
|
|
if(now - _lastReceived > _packetTimeOutMS)
|
|
{
|
|
packetTimeOut = true;
|
|
_lastReceived = 0; // only one callback
|
|
}
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}
|
|
CriticalSectionScoped lock(_criticalSectionFeedbacks);
|
|
if(packetTimeOut && _cbRtcpFeedback)
|
|
{
|
|
_cbRtcpFeedback->OnRTCPPacketTimeout(_id);
|
|
}
|
|
}
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} // namespace webrtc
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